Refactor AudioPolicyManager
AudioPolicyManager implementation is now split into the
following files:
files managerdefault/Gains.*
class AudioGain
class VolumeCurvePoint
class StreamDescriptor
files managerdefault/Devices.*
class DeviceDescriptor
class DeviceVector
files managerdefault/Ports.*
class AudioPort
class AudioPortConfig
class AudioPatch
files managerdefault/IOProfile.*
class IOProfile
files managerdefault/HwModule.*
class HwModule
files managerdefault/AudioInputDescriptor.*
class AudioInputDescriptor
files managerdefault/AudioOutputDescriptor.*
class AudioOutputDescriptor
All files for libaudiopolicyservice are moved under service/
All files for libaudiopolicymanager are moved under manager/
Change-Id: I43758be1894e37d34db194b51a19ae24461e066e
diff --git a/services/audiopolicy/managerdefault/AudioPolicyManager.cpp b/services/audiopolicy/managerdefault/AudioPolicyManager.cpp
new file mode 100644
index 0000000..b48dc80
--- /dev/null
+++ b/services/audiopolicy/managerdefault/AudioPolicyManager.cpp
@@ -0,0 +1,5766 @@
+/*
+ * Copyright (C) 2009 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#define LOG_TAG "APM::AudioPolicyManager"
+//#define LOG_NDEBUG 0
+
+//#define VERY_VERBOSE_LOGGING
+#ifdef VERY_VERBOSE_LOGGING
+#define ALOGVV ALOGV
+#else
+#define ALOGVV(a...) do { } while(0)
+#endif
+
+// A device mask for all audio input devices that are considered "virtual" when evaluating
+// active inputs in getActiveInput()
+#define APM_AUDIO_IN_DEVICE_VIRTUAL_ALL (AUDIO_DEVICE_IN_REMOTE_SUBMIX|AUDIO_DEVICE_IN_FM_TUNER)
+// A device mask for all audio output devices that are considered "remote" when evaluating
+// active output devices in isStreamActiveRemotely()
+#define APM_AUDIO_OUT_DEVICE_REMOTE_ALL AUDIO_DEVICE_OUT_REMOTE_SUBMIX
+// A device mask for all audio input and output devices where matching inputs/outputs on device
+// type alone is not enough: the address must match too
+#define APM_AUDIO_DEVICE_MATCH_ADDRESS_ALL (AUDIO_DEVICE_IN_REMOTE_SUBMIX | \
+ AUDIO_DEVICE_OUT_REMOTE_SUBMIX)
+
+#include <inttypes.h>
+#include <math.h>
+
+#include <cutils/properties.h>
+#include <utils/Log.h>
+#include <hardware/audio.h>
+#include <hardware/audio_effect.h>
+#include <media/AudioParameter.h>
+#include <media/AudioPolicyHelper.h>
+#include <soundtrigger/SoundTrigger.h>
+#include "AudioPolicyManager.h"
+#include "audio_policy_conf.h"
+
+namespace android {
+
+// ----------------------------------------------------------------------------
+// AudioPolicyInterface implementation
+// ----------------------------------------------------------------------------
+
+status_t AudioPolicyManager::setDeviceConnectionState(audio_devices_t device,
+ audio_policy_dev_state_t state,
+ const char *device_address,
+ const char *device_name)
+{
+ return setDeviceConnectionStateInt(device, state, device_address, device_name);
+}
+
+status_t AudioPolicyManager::setDeviceConnectionStateInt(audio_devices_t device,
+ audio_policy_dev_state_t state,
+ const char *device_address,
+ const char *device_name)
+{
+ ALOGV("setDeviceConnectionStateInt() device: 0x%X, state %d, address %s name %s",
+- device, state, device_address, device_name);
+
+ // connect/disconnect only 1 device at a time
+ if (!audio_is_output_device(device) && !audio_is_input_device(device)) return BAD_VALUE;
+
+ sp<DeviceDescriptor> devDesc = getDeviceDescriptor(device, device_address, device_name);
+
+ // handle output devices
+ if (audio_is_output_device(device)) {
+ SortedVector <audio_io_handle_t> outputs;
+
+ ssize_t index = mAvailableOutputDevices.indexOf(devDesc);
+
+ // save a copy of the opened output descriptors before any output is opened or closed
+ // by checkOutputsForDevice(). This will be needed by checkOutputForAllStrategies()
+ mPreviousOutputs = mOutputs;
+ switch (state)
+ {
+ // handle output device connection
+ case AUDIO_POLICY_DEVICE_STATE_AVAILABLE: {
+ if (index >= 0) {
+ ALOGW("setDeviceConnectionState() device already connected: %x", device);
+ return INVALID_OPERATION;
+ }
+ ALOGV("setDeviceConnectionState() connecting device %x", device);
+
+ // register new device as available
+ index = mAvailableOutputDevices.add(devDesc);
+ if (index >= 0) {
+ sp<HwModule> module = getModuleForDevice(device);
+ if (module == 0) {
+ ALOGD("setDeviceConnectionState() could not find HW module for device %08x",
+ device);
+ mAvailableOutputDevices.remove(devDesc);
+ return INVALID_OPERATION;
+ }
+ mAvailableOutputDevices[index]->attach(module);
+ } else {
+ return NO_MEMORY;
+ }
+
+ if (checkOutputsForDevice(devDesc, state, outputs, devDesc->mAddress) != NO_ERROR) {
+ mAvailableOutputDevices.remove(devDesc);
+ return INVALID_OPERATION;
+ }
+ // outputs should never be empty here
+ ALOG_ASSERT(outputs.size() != 0, "setDeviceConnectionState():"
+ "checkOutputsForDevice() returned no outputs but status OK");
+ ALOGV("setDeviceConnectionState() checkOutputsForDevice() returned %zu outputs",
+ outputs.size());
+
+ // Send connect to HALs
+ AudioParameter param = AudioParameter(devDesc->mAddress);
+ param.addInt(String8(AUDIO_PARAMETER_DEVICE_CONNECT), device);
+ mpClientInterface->setParameters(AUDIO_IO_HANDLE_NONE, param.toString());
+
+ } break;
+ // handle output device disconnection
+ case AUDIO_POLICY_DEVICE_STATE_UNAVAILABLE: {
+ if (index < 0) {
+ ALOGW("setDeviceConnectionState() device not connected: %x", device);
+ return INVALID_OPERATION;
+ }
+
+ ALOGV("setDeviceConnectionState() disconnecting output device %x", device);
+
+ // Send Disconnect to HALs
+ AudioParameter param = AudioParameter(devDesc->mAddress);
+ param.addInt(String8(AUDIO_PARAMETER_DEVICE_DISCONNECT), device);
+ mpClientInterface->setParameters(AUDIO_IO_HANDLE_NONE, param.toString());
+
+ // remove device from available output devices
+ mAvailableOutputDevices.remove(devDesc);
+
+ checkOutputsForDevice(devDesc, state, outputs, devDesc->mAddress);
+ } break;
+
+ default:
+ ALOGE("setDeviceConnectionState() invalid state: %x", state);
+ return BAD_VALUE;
+ }
+
+ // checkA2dpSuspend must run before checkOutputForAllStrategies so that A2DP
+ // output is suspended before any tracks are moved to it
+ checkA2dpSuspend();
+ checkOutputForAllStrategies();
+ // outputs must be closed after checkOutputForAllStrategies() is executed
+ if (!outputs.isEmpty()) {
+ for (size_t i = 0; i < outputs.size(); i++) {
+ sp<AudioOutputDescriptor> desc = mOutputs.valueFor(outputs[i]);
+ // close unused outputs after device disconnection or direct outputs that have been
+ // opened by checkOutputsForDevice() to query dynamic parameters
+ if ((state == AUDIO_POLICY_DEVICE_STATE_UNAVAILABLE) ||
+ (((desc->mFlags & AUDIO_OUTPUT_FLAG_DIRECT) != 0) &&
+ (desc->mDirectOpenCount == 0))) {
+ closeOutput(outputs[i]);
+ }
+ }
+ // check again after closing A2DP output to reset mA2dpSuspended if needed
+ checkA2dpSuspend();
+ }
+
+ updateDevicesAndOutputs();
+ if (mPhoneState == AUDIO_MODE_IN_CALL) {
+ audio_devices_t newDevice = getNewOutputDevice(mPrimaryOutput, false /*fromCache*/);
+ updateCallRouting(newDevice);
+ }
+ for (size_t i = 0; i < mOutputs.size(); i++) {
+ audio_io_handle_t output = mOutputs.keyAt(i);
+ if ((mPhoneState != AUDIO_MODE_IN_CALL) || (output != mPrimaryOutput)) {
+ audio_devices_t newDevice = getNewOutputDevice(mOutputs.keyAt(i),
+ true /*fromCache*/);
+ // do not force device change on duplicated output because if device is 0, it will
+ // also force a device 0 for the two outputs it is duplicated to which may override
+ // a valid device selection on those outputs.
+ bool force = !mOutputs.valueAt(i)->isDuplicated()
+ && (!deviceDistinguishesOnAddress(device)
+ // always force when disconnecting (a non-duplicated device)
+ || (state == AUDIO_POLICY_DEVICE_STATE_UNAVAILABLE));
+ setOutputDevice(output, newDevice, force, 0);
+ }
+ }
+
+ mpClientInterface->onAudioPortListUpdate();
+ return NO_ERROR;
+ } // end if is output device
+
+ // handle input devices
+ if (audio_is_input_device(device)) {
+ SortedVector <audio_io_handle_t> inputs;
+
+ ssize_t index = mAvailableInputDevices.indexOf(devDesc);
+ switch (state)
+ {
+ // handle input device connection
+ case AUDIO_POLICY_DEVICE_STATE_AVAILABLE: {
+ if (index >= 0) {
+ ALOGW("setDeviceConnectionState() device already connected: %d", device);
+ return INVALID_OPERATION;
+ }
+ sp<HwModule> module = getModuleForDevice(device);
+ if (module == NULL) {
+ ALOGW("setDeviceConnectionState(): could not find HW module for device %08x",
+ device);
+ return INVALID_OPERATION;
+ }
+ if (checkInputsForDevice(device, state, inputs, devDesc->mAddress) != NO_ERROR) {
+ return INVALID_OPERATION;
+ }
+
+ index = mAvailableInputDevices.add(devDesc);
+ if (index >= 0) {
+ mAvailableInputDevices[index]->attach(module);
+ } else {
+ return NO_MEMORY;
+ }
+
+ // Set connect to HALs
+ AudioParameter param = AudioParameter(devDesc->mAddress);
+ param.addInt(String8(AUDIO_PARAMETER_DEVICE_CONNECT), device);
+ mpClientInterface->setParameters(AUDIO_IO_HANDLE_NONE, param.toString());
+
+ } break;
+
+ // handle input device disconnection
+ case AUDIO_POLICY_DEVICE_STATE_UNAVAILABLE: {
+ if (index < 0) {
+ ALOGW("setDeviceConnectionState() device not connected: %d", device);
+ return INVALID_OPERATION;
+ }
+
+ ALOGV("setDeviceConnectionState() disconnecting input device %x", device);
+
+ // Set Disconnect to HALs
+ AudioParameter param = AudioParameter(devDesc->mAddress);
+ param.addInt(String8(AUDIO_PARAMETER_DEVICE_DISCONNECT), device);
+ mpClientInterface->setParameters(AUDIO_IO_HANDLE_NONE, param.toString());
+
+ checkInputsForDevice(device, state, inputs, devDesc->mAddress);
+ mAvailableInputDevices.remove(devDesc);
+
+ } break;
+
+ default:
+ ALOGE("setDeviceConnectionState() invalid state: %x", state);
+ return BAD_VALUE;
+ }
+
+ closeAllInputs();
+
+ if (mPhoneState == AUDIO_MODE_IN_CALL) {
+ audio_devices_t newDevice = getNewOutputDevice(mPrimaryOutput, false /*fromCache*/);
+ updateCallRouting(newDevice);
+ }
+
+ mpClientInterface->onAudioPortListUpdate();
+ return NO_ERROR;
+ } // end if is input device
+
+ ALOGW("setDeviceConnectionState() invalid device: %x", device);
+ return BAD_VALUE;
+}
+
+audio_policy_dev_state_t AudioPolicyManager::getDeviceConnectionState(audio_devices_t device,
+ const char *device_address)
+{
+ sp<DeviceDescriptor> devDesc = getDeviceDescriptor(device, device_address, "");
+ DeviceVector *deviceVector;
+
+ if (audio_is_output_device(device)) {
+ deviceVector = &mAvailableOutputDevices;
+ } else if (audio_is_input_device(device)) {
+ deviceVector = &mAvailableInputDevices;
+ } else {
+ ALOGW("getDeviceConnectionState() invalid device type %08x", device);
+ return AUDIO_POLICY_DEVICE_STATE_UNAVAILABLE;
+ }
+
+ ssize_t index = deviceVector->indexOf(devDesc);
+ if (index >= 0) {
+ return AUDIO_POLICY_DEVICE_STATE_AVAILABLE;
+ } else {
+ return AUDIO_POLICY_DEVICE_STATE_UNAVAILABLE;
+ }
+}
+
+sp<DeviceDescriptor> AudioPolicyManager::getDeviceDescriptor(const audio_devices_t device,
+ const char *device_address,
+ const char *device_name)
+{
+ String8 address = (device_address == NULL) ? String8("") : String8(device_address);
+ // handle legacy remote submix case where the address was not always specified
+ if (deviceDistinguishesOnAddress(device) && (address.length() == 0)) {
+ address = String8("0");
+ }
+
+ for (size_t i = 0; i < mHwModules.size(); i++) {
+ if (mHwModules[i]->mHandle == 0) {
+ continue;
+ }
+ DeviceVector deviceList =
+ mHwModules[i]->mDeclaredDevices.getDevicesFromTypeAddr(device, address);
+ if (!deviceList.isEmpty()) {
+ return deviceList.itemAt(0);
+ }
+ deviceList = mHwModules[i]->mDeclaredDevices.getDevicesFromType(device);
+ if (!deviceList.isEmpty()) {
+ return deviceList.itemAt(0);
+ }
+ }
+
+ sp<DeviceDescriptor> devDesc =
+ new DeviceDescriptor(String8(device_name != NULL ? device_name : ""), device);
+ devDesc->mAddress = address;
+ return devDesc;
+}
+
+void AudioPolicyManager::updateCallRouting(audio_devices_t rxDevice, int delayMs)
+{
+ bool createTxPatch = false;
+ struct audio_patch patch;
+ patch.num_sources = 1;
+ patch.num_sinks = 1;
+ status_t status;
+ audio_patch_handle_t afPatchHandle;
+ DeviceVector deviceList;
+
+ audio_devices_t txDevice = getDeviceAndMixForInputSource(AUDIO_SOURCE_VOICE_COMMUNICATION);
+ ALOGV("updateCallRouting device rxDevice %08x txDevice %08x", rxDevice, txDevice);
+
+ // release existing RX patch if any
+ if (mCallRxPatch != 0) {
+ mpClientInterface->releaseAudioPatch(mCallRxPatch->mAfPatchHandle, 0);
+ mCallRxPatch.clear();
+ }
+ // release TX patch if any
+ if (mCallTxPatch != 0) {
+ mpClientInterface->releaseAudioPatch(mCallTxPatch->mAfPatchHandle, 0);
+ mCallTxPatch.clear();
+ }
+
+ // If the RX device is on the primary HW module, then use legacy routing method for voice calls
+ // via setOutputDevice() on primary output.
+ // Otherwise, create two audio patches for TX and RX path.
+ if (availablePrimaryOutputDevices() & rxDevice) {
+ setOutputDevice(mPrimaryOutput, rxDevice, true, delayMs);
+ // If the TX device is also on the primary HW module, setOutputDevice() will take care
+ // of it due to legacy implementation. If not, create a patch.
+ if ((availablePrimaryInputDevices() & txDevice & ~AUDIO_DEVICE_BIT_IN)
+ == AUDIO_DEVICE_NONE) {
+ createTxPatch = true;
+ }
+ } else {
+ // create RX path audio patch
+ deviceList = mAvailableOutputDevices.getDevicesFromType(rxDevice);
+ ALOG_ASSERT(!deviceList.isEmpty(),
+ "updateCallRouting() selected device not in output device list");
+ sp<DeviceDescriptor> rxSinkDeviceDesc = deviceList.itemAt(0);
+ deviceList = mAvailableInputDevices.getDevicesFromType(AUDIO_DEVICE_IN_TELEPHONY_RX);
+ ALOG_ASSERT(!deviceList.isEmpty(),
+ "updateCallRouting() no telephony RX device");
+ sp<DeviceDescriptor> rxSourceDeviceDesc = deviceList.itemAt(0);
+
+ rxSourceDeviceDesc->toAudioPortConfig(&patch.sources[0]);
+ rxSinkDeviceDesc->toAudioPortConfig(&patch.sinks[0]);
+
+ // request to reuse existing output stream if one is already opened to reach the RX device
+ SortedVector<audio_io_handle_t> outputs =
+ getOutputsForDevice(rxDevice, mOutputs);
+ audio_io_handle_t output = selectOutput(outputs,
+ AUDIO_OUTPUT_FLAG_NONE,
+ AUDIO_FORMAT_INVALID);
+ if (output != AUDIO_IO_HANDLE_NONE) {
+ sp<AudioOutputDescriptor> outputDesc = mOutputs.valueFor(output);
+ ALOG_ASSERT(!outputDesc->isDuplicated(),
+ "updateCallRouting() RX device output is duplicated");
+ outputDesc->toAudioPortConfig(&patch.sources[1]);
+ patch.num_sources = 2;
+ }
+
+ afPatchHandle = AUDIO_PATCH_HANDLE_NONE;
+ status = mpClientInterface->createAudioPatch(&patch, &afPatchHandle, 0);
+ ALOGW_IF(status != NO_ERROR, "updateCallRouting() error %d creating RX audio patch",
+ status);
+ if (status == NO_ERROR) {
+ mCallRxPatch = new AudioPatch((audio_patch_handle_t)nextUniqueId(),
+ &patch, mUidCached);
+ mCallRxPatch->mAfPatchHandle = afPatchHandle;
+ mCallRxPatch->mUid = mUidCached;
+ }
+ createTxPatch = true;
+ }
+ if (createTxPatch) {
+
+ struct audio_patch patch;
+ patch.num_sources = 1;
+ patch.num_sinks = 1;
+ deviceList = mAvailableInputDevices.getDevicesFromType(txDevice);
+ ALOG_ASSERT(!deviceList.isEmpty(),
+ "updateCallRouting() selected device not in input device list");
+ sp<DeviceDescriptor> txSourceDeviceDesc = deviceList.itemAt(0);
+ txSourceDeviceDesc->toAudioPortConfig(&patch.sources[0]);
+ deviceList = mAvailableOutputDevices.getDevicesFromType(AUDIO_DEVICE_OUT_TELEPHONY_TX);
+ ALOG_ASSERT(!deviceList.isEmpty(),
+ "updateCallRouting() no telephony TX device");
+ sp<DeviceDescriptor> txSinkDeviceDesc = deviceList.itemAt(0);
+ txSinkDeviceDesc->toAudioPortConfig(&patch.sinks[0]);
+
+ SortedVector<audio_io_handle_t> outputs =
+ getOutputsForDevice(AUDIO_DEVICE_OUT_TELEPHONY_TX, mOutputs);
+ audio_io_handle_t output = selectOutput(outputs,
+ AUDIO_OUTPUT_FLAG_NONE,
+ AUDIO_FORMAT_INVALID);
+ // request to reuse existing output stream if one is already opened to reach the TX
+ // path output device
+ if (output != AUDIO_IO_HANDLE_NONE) {
+ sp<AudioOutputDescriptor> outputDesc = mOutputs.valueFor(output);
+ ALOG_ASSERT(!outputDesc->isDuplicated(),
+ "updateCallRouting() RX device output is duplicated");
+ outputDesc->toAudioPortConfig(&patch.sources[1]);
+ patch.num_sources = 2;
+ }
+
+ afPatchHandle = AUDIO_PATCH_HANDLE_NONE;
+ status = mpClientInterface->createAudioPatch(&patch, &afPatchHandle, 0);
+ ALOGW_IF(status != NO_ERROR, "setPhoneState() error %d creating TX audio patch",
+ status);
+ if (status == NO_ERROR) {
+ mCallTxPatch = new AudioPatch((audio_patch_handle_t)nextUniqueId(),
+ &patch, mUidCached);
+ mCallTxPatch->mAfPatchHandle = afPatchHandle;
+ mCallTxPatch->mUid = mUidCached;
+ }
+ }
+}
+
+void AudioPolicyManager::setPhoneState(audio_mode_t state)
+{
+ ALOGV("setPhoneState() state %d", state);
+ if (state < 0 || state >= AUDIO_MODE_CNT) {
+ ALOGW("setPhoneState() invalid state %d", state);
+ return;
+ }
+
+ if (state == mPhoneState ) {
+ ALOGW("setPhoneState() setting same state %d", state);
+ return;
+ }
+
+ // if leaving call state, handle special case of active streams
+ // pertaining to sonification strategy see handleIncallSonification()
+ if (isInCall()) {
+ ALOGV("setPhoneState() in call state management: new state is %d", state);
+ for (int stream = 0; stream < AUDIO_STREAM_CNT; stream++) {
+ if (stream == AUDIO_STREAM_PATCH) {
+ continue;
+ }
+ handleIncallSonification((audio_stream_type_t)stream, false, true);
+ }
+
+ // force reevaluating accessibility routing when call starts
+ mpClientInterface->invalidateStream(AUDIO_STREAM_ACCESSIBILITY);
+ }
+
+ // store previous phone state for management of sonification strategy below
+ int oldState = mPhoneState;
+ mPhoneState = state;
+ bool force = false;
+
+ // are we entering or starting a call
+ if (!isStateInCall(oldState) && isStateInCall(state)) {
+ ALOGV(" Entering call in setPhoneState()");
+ // force routing command to audio hardware when starting a call
+ // even if no device change is needed
+ force = true;
+ for (int j = 0; j < ApmGains::DEVICE_CATEGORY_CNT; j++) {
+ mStreams[AUDIO_STREAM_DTMF].mVolumeCurve[j] =
+ ApmGains::sVolumeProfiles[AUDIO_STREAM_VOICE_CALL][j];
+ }
+ } else if (isStateInCall(oldState) && !isStateInCall(state)) {
+ ALOGV(" Exiting call in setPhoneState()");
+ // force routing command to audio hardware when exiting a call
+ // even if no device change is needed
+ force = true;
+ for (int j = 0; j < ApmGains::DEVICE_CATEGORY_CNT; j++) {
+ mStreams[AUDIO_STREAM_DTMF].mVolumeCurve[j] =
+ ApmGains::sVolumeProfiles[AUDIO_STREAM_DTMF][j];
+ }
+ } else if (isStateInCall(state) && (state != oldState)) {
+ ALOGV(" Switching between telephony and VoIP in setPhoneState()");
+ // force routing command to audio hardware when switching between telephony and VoIP
+ // even if no device change is needed
+ force = true;
+ }
+
+ // check for device and output changes triggered by new phone state
+ checkA2dpSuspend();
+ checkOutputForAllStrategies();
+ updateDevicesAndOutputs();
+
+ sp<AudioOutputDescriptor> hwOutputDesc = mOutputs.valueFor(mPrimaryOutput);
+
+ int delayMs = 0;
+ if (isStateInCall(state)) {
+ nsecs_t sysTime = systemTime();
+ for (size_t i = 0; i < mOutputs.size(); i++) {
+ sp<AudioOutputDescriptor> desc = mOutputs.valueAt(i);
+ // mute media and sonification strategies and delay device switch by the largest
+ // latency of any output where either strategy is active.
+ // This avoid sending the ring tone or music tail into the earpiece or headset.
+ if ((desc->isStrategyActive(STRATEGY_MEDIA,
+ SONIFICATION_HEADSET_MUSIC_DELAY,
+ sysTime) ||
+ desc->isStrategyActive(STRATEGY_SONIFICATION,
+ SONIFICATION_HEADSET_MUSIC_DELAY,
+ sysTime)) &&
+ (delayMs < (int)desc->mLatency*2)) {
+ delayMs = desc->mLatency*2;
+ }
+ setStrategyMute(STRATEGY_MEDIA, true, mOutputs.keyAt(i));
+ setStrategyMute(STRATEGY_MEDIA, false, mOutputs.keyAt(i), MUTE_TIME_MS,
+ getDeviceForStrategy(STRATEGY_MEDIA, true /*fromCache*/));
+ setStrategyMute(STRATEGY_SONIFICATION, true, mOutputs.keyAt(i));
+ setStrategyMute(STRATEGY_SONIFICATION, false, mOutputs.keyAt(i), MUTE_TIME_MS,
+ getDeviceForStrategy(STRATEGY_SONIFICATION, true /*fromCache*/));
+ }
+ }
+
+ // Note that despite the fact that getNewOutputDevice() is called on the primary output,
+ // the device returned is not necessarily reachable via this output
+ audio_devices_t rxDevice = getNewOutputDevice(mPrimaryOutput, false /*fromCache*/);
+ // force routing command to audio hardware when ending call
+ // even if no device change is needed
+ if (isStateInCall(oldState) && rxDevice == AUDIO_DEVICE_NONE) {
+ rxDevice = hwOutputDesc->device();
+ }
+
+ if (state == AUDIO_MODE_IN_CALL) {
+ updateCallRouting(rxDevice, delayMs);
+ } else if (oldState == AUDIO_MODE_IN_CALL) {
+ if (mCallRxPatch != 0) {
+ mpClientInterface->releaseAudioPatch(mCallRxPatch->mAfPatchHandle, 0);
+ mCallRxPatch.clear();
+ }
+ if (mCallTxPatch != 0) {
+ mpClientInterface->releaseAudioPatch(mCallTxPatch->mAfPatchHandle, 0);
+ mCallTxPatch.clear();
+ }
+ setOutputDevice(mPrimaryOutput, rxDevice, force, 0);
+ } else {
+ setOutputDevice(mPrimaryOutput, rxDevice, force, 0);
+ }
+ // if entering in call state, handle special case of active streams
+ // pertaining to sonification strategy see handleIncallSonification()
+ if (isStateInCall(state)) {
+ ALOGV("setPhoneState() in call state management: new state is %d", state);
+ for (int stream = 0; stream < AUDIO_STREAM_CNT; stream++) {
+ if (stream == AUDIO_STREAM_PATCH) {
+ continue;
+ }
+ handleIncallSonification((audio_stream_type_t)stream, true, true);
+ }
+ }
+
+ // Flag that ringtone volume must be limited to music volume until we exit MODE_RINGTONE
+ if (state == AUDIO_MODE_RINGTONE &&
+ isStreamActive(AUDIO_STREAM_MUSIC, SONIFICATION_HEADSET_MUSIC_DELAY)) {
+ mLimitRingtoneVolume = true;
+ } else {
+ mLimitRingtoneVolume = false;
+ }
+}
+
+void AudioPolicyManager::setForceUse(audio_policy_force_use_t usage,
+ audio_policy_forced_cfg_t config)
+{
+ ALOGV("setForceUse() usage %d, config %d, mPhoneState %d", usage, config, mPhoneState);
+
+ bool forceVolumeReeval = false;
+ switch(usage) {
+ case AUDIO_POLICY_FORCE_FOR_COMMUNICATION:
+ if (config != AUDIO_POLICY_FORCE_SPEAKER && config != AUDIO_POLICY_FORCE_BT_SCO &&
+ config != AUDIO_POLICY_FORCE_NONE) {
+ ALOGW("setForceUse() invalid config %d for FOR_COMMUNICATION", config);
+ return;
+ }
+ forceVolumeReeval = true;
+ mForceUse[usage] = config;
+ break;
+ case AUDIO_POLICY_FORCE_FOR_MEDIA:
+ if (config != AUDIO_POLICY_FORCE_HEADPHONES && config != AUDIO_POLICY_FORCE_BT_A2DP &&
+ config != AUDIO_POLICY_FORCE_WIRED_ACCESSORY &&
+ config != AUDIO_POLICY_FORCE_ANALOG_DOCK &&
+ config != AUDIO_POLICY_FORCE_DIGITAL_DOCK && config != AUDIO_POLICY_FORCE_NONE &&
+ config != AUDIO_POLICY_FORCE_NO_BT_A2DP && config != AUDIO_POLICY_FORCE_SPEAKER ) {
+ ALOGW("setForceUse() invalid config %d for FOR_MEDIA", config);
+ return;
+ }
+ mForceUse[usage] = config;
+ break;
+ case AUDIO_POLICY_FORCE_FOR_RECORD:
+ if (config != AUDIO_POLICY_FORCE_BT_SCO && config != AUDIO_POLICY_FORCE_WIRED_ACCESSORY &&
+ config != AUDIO_POLICY_FORCE_NONE) {
+ ALOGW("setForceUse() invalid config %d for FOR_RECORD", config);
+ return;
+ }
+ mForceUse[usage] = config;
+ break;
+ case AUDIO_POLICY_FORCE_FOR_DOCK:
+ if (config != AUDIO_POLICY_FORCE_NONE && config != AUDIO_POLICY_FORCE_BT_CAR_DOCK &&
+ config != AUDIO_POLICY_FORCE_BT_DESK_DOCK &&
+ config != AUDIO_POLICY_FORCE_WIRED_ACCESSORY &&
+ config != AUDIO_POLICY_FORCE_ANALOG_DOCK &&
+ config != AUDIO_POLICY_FORCE_DIGITAL_DOCK) {
+ ALOGW("setForceUse() invalid config %d for FOR_DOCK", config);
+ }
+ forceVolumeReeval = true;
+ mForceUse[usage] = config;
+ break;
+ case AUDIO_POLICY_FORCE_FOR_SYSTEM:
+ if (config != AUDIO_POLICY_FORCE_NONE &&
+ config != AUDIO_POLICY_FORCE_SYSTEM_ENFORCED) {
+ ALOGW("setForceUse() invalid config %d for FOR_SYSTEM", config);
+ }
+ forceVolumeReeval = true;
+ mForceUse[usage] = config;
+ break;
+ case AUDIO_POLICY_FORCE_FOR_HDMI_SYSTEM_AUDIO:
+ if (config != AUDIO_POLICY_FORCE_NONE &&
+ config != AUDIO_POLICY_FORCE_HDMI_SYSTEM_AUDIO_ENFORCED) {
+ ALOGW("setForceUse() invalid config %d forHDMI_SYSTEM_AUDIO", config);
+ }
+ mForceUse[usage] = config;
+ break;
+ default:
+ ALOGW("setForceUse() invalid usage %d", usage);
+ break;
+ }
+
+ // check for device and output changes triggered by new force usage
+ checkA2dpSuspend();
+ checkOutputForAllStrategies();
+ updateDevicesAndOutputs();
+ if (mPhoneState == AUDIO_MODE_IN_CALL) {
+ audio_devices_t newDevice = getNewOutputDevice(mPrimaryOutput, true /*fromCache*/);
+ updateCallRouting(newDevice);
+ }
+ for (size_t i = 0; i < mOutputs.size(); i++) {
+ audio_io_handle_t output = mOutputs.keyAt(i);
+ audio_devices_t newDevice = getNewOutputDevice(output, true /*fromCache*/);
+ if ((mPhoneState != AUDIO_MODE_IN_CALL) || (output != mPrimaryOutput)) {
+ setOutputDevice(output, newDevice, (newDevice != AUDIO_DEVICE_NONE));
+ }
+ if (forceVolumeReeval && (newDevice != AUDIO_DEVICE_NONE)) {
+ applyStreamVolumes(output, newDevice, 0, true);
+ }
+ }
+
+ audio_io_handle_t activeInput = getActiveInput();
+ if (activeInput != 0) {
+ setInputDevice(activeInput, getNewInputDevice(activeInput));
+ }
+
+}
+
+audio_policy_forced_cfg_t AudioPolicyManager::getForceUse(audio_policy_force_use_t usage)
+{
+ return mForceUse[usage];
+}
+
+void AudioPolicyManager::setSystemProperty(const char* property, const char* value)
+{
+ ALOGV("setSystemProperty() property %s, value %s", property, value);
+}
+
+// Find a direct output profile compatible with the parameters passed, even if the input flags do
+// not explicitly request a direct output
+sp<IOProfile> AudioPolicyManager::getProfileForDirectOutput(
+ audio_devices_t device,
+ uint32_t samplingRate,
+ audio_format_t format,
+ audio_channel_mask_t channelMask,
+ audio_output_flags_t flags)
+{
+ for (size_t i = 0; i < mHwModules.size(); i++) {
+ if (mHwModules[i]->mHandle == 0) {
+ continue;
+ }
+ for (size_t j = 0; j < mHwModules[i]->mOutputProfiles.size(); j++) {
+ sp<IOProfile> profile = mHwModules[i]->mOutputProfiles[j];
+ bool found = profile->isCompatibleProfile(device, String8(""), samplingRate,
+ NULL /*updatedSamplingRate*/, format, channelMask,
+ flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD ?
+ AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD : AUDIO_OUTPUT_FLAG_DIRECT);
+ if (found && (mAvailableOutputDevices.types() & profile->mSupportedDevices.types())) {
+ return profile;
+ }
+ }
+ }
+ return 0;
+}
+
+audio_io_handle_t AudioPolicyManager::getOutput(audio_stream_type_t stream,
+ uint32_t samplingRate,
+ audio_format_t format,
+ audio_channel_mask_t channelMask,
+ audio_output_flags_t flags,
+ const audio_offload_info_t *offloadInfo)
+{
+ routing_strategy strategy = getStrategy(stream);
+ audio_devices_t device = getDeviceForStrategy(strategy, false /*fromCache*/);
+ ALOGV("getOutput() device %d, stream %d, samplingRate %d, format %x, channelMask %x, flags %x",
+ device, stream, samplingRate, format, channelMask, flags);
+
+ return getOutputForDevice(device, AUDIO_SESSION_ALLOCATE,
+ stream, samplingRate,format, channelMask,
+ flags, offloadInfo);
+}
+
+status_t AudioPolicyManager::getOutputForAttr(const audio_attributes_t *attr,
+ audio_io_handle_t *output,
+ audio_session_t session,
+ audio_stream_type_t *stream,
+ uint32_t samplingRate,
+ audio_format_t format,
+ audio_channel_mask_t channelMask,
+ audio_output_flags_t flags,
+ const audio_offload_info_t *offloadInfo)
+{
+ audio_attributes_t attributes;
+ if (attr != NULL) {
+ if (!isValidAttributes(attr)) {
+ ALOGE("getOutputForAttr() invalid attributes: usage=%d content=%d flags=0x%x tags=[%s]",
+ attr->usage, attr->content_type, attr->flags,
+ attr->tags);
+ return BAD_VALUE;
+ }
+ attributes = *attr;
+ } else {
+ if (*stream < AUDIO_STREAM_MIN || *stream >= AUDIO_STREAM_PUBLIC_CNT) {
+ ALOGE("getOutputForAttr(): invalid stream type");
+ return BAD_VALUE;
+ }
+ stream_type_to_audio_attributes(*stream, &attributes);
+ }
+
+ for (size_t i = 0; i < mPolicyMixes.size(); i++) {
+ sp<AudioOutputDescriptor> desc;
+ if (mPolicyMixes[i]->mMix.mMixType == MIX_TYPE_PLAYERS) {
+ for (size_t j = 0; j < mPolicyMixes[i]->mMix.mCriteria.size(); j++) {
+ if ((RULE_MATCH_ATTRIBUTE_USAGE == mPolicyMixes[i]->mMix.mCriteria[j].mRule &&
+ mPolicyMixes[i]->mMix.mCriteria[j].mAttr.mUsage == attributes.usage) ||
+ (RULE_EXCLUDE_ATTRIBUTE_USAGE == mPolicyMixes[i]->mMix.mCriteria[j].mRule &&
+ mPolicyMixes[i]->mMix.mCriteria[j].mAttr.mUsage != attributes.usage)) {
+ desc = mPolicyMixes[i]->mOutput;
+ break;
+ }
+ if (strncmp(attributes.tags, "addr=", strlen("addr=")) == 0 &&
+ strncmp(attributes.tags + strlen("addr="),
+ mPolicyMixes[i]->mMix.mRegistrationId.string(),
+ AUDIO_ATTRIBUTES_TAGS_MAX_SIZE - strlen("addr=") - 1) == 0) {
+ desc = mPolicyMixes[i]->mOutput;
+ break;
+ }
+ }
+ } else if (mPolicyMixes[i]->mMix.mMixType == MIX_TYPE_RECORDERS) {
+ if (attributes.usage == AUDIO_USAGE_VIRTUAL_SOURCE &&
+ strncmp(attributes.tags, "addr=", strlen("addr=")) == 0 &&
+ strncmp(attributes.tags + strlen("addr="),
+ mPolicyMixes[i]->mMix.mRegistrationId.string(),
+ AUDIO_ATTRIBUTES_TAGS_MAX_SIZE - strlen("addr=") - 1) == 0) {
+ desc = mPolicyMixes[i]->mOutput;
+ }
+ }
+ if (desc != 0) {
+ if (!audio_is_linear_pcm(format)) {
+ return BAD_VALUE;
+ }
+ desc->mPolicyMix = &mPolicyMixes[i]->mMix;
+ *stream = streamTypefromAttributesInt(&attributes);
+ *output = desc->mIoHandle;
+ ALOGV("getOutputForAttr() returns output %d", *output);
+ return NO_ERROR;
+ }
+ }
+ if (attributes.usage == AUDIO_USAGE_VIRTUAL_SOURCE) {
+ ALOGW("getOutputForAttr() no policy mix found for usage AUDIO_USAGE_VIRTUAL_SOURCE");
+ return BAD_VALUE;
+ }
+
+ ALOGV("getOutputForAttr() usage=%d, content=%d, tag=%s flags=%08x",
+ attributes.usage, attributes.content_type, attributes.tags, attributes.flags);
+
+ routing_strategy strategy = (routing_strategy) getStrategyForAttr(&attributes);
+ audio_devices_t device = getDeviceForStrategy(strategy, false /*fromCache*/);
+
+ if ((attributes.flags & AUDIO_FLAG_HW_AV_SYNC) != 0) {
+ flags = (audio_output_flags_t)(flags | AUDIO_OUTPUT_FLAG_HW_AV_SYNC);
+ }
+
+ ALOGV("getOutputForAttr() device 0x%x, samplingRate %d, format %x, channelMask %x, flags %x",
+ device, samplingRate, format, channelMask, flags);
+
+ *stream = streamTypefromAttributesInt(&attributes);
+ *output = getOutputForDevice(device, session, *stream,
+ samplingRate, format, channelMask,
+ flags, offloadInfo);
+ if (*output == AUDIO_IO_HANDLE_NONE) {
+ return INVALID_OPERATION;
+ }
+ return NO_ERROR;
+}
+
+audio_io_handle_t AudioPolicyManager::getOutputForDevice(
+ audio_devices_t device,
+ audio_session_t session __unused,
+ audio_stream_type_t stream,
+ uint32_t samplingRate,
+ audio_format_t format,
+ audio_channel_mask_t channelMask,
+ audio_output_flags_t flags,
+ const audio_offload_info_t *offloadInfo)
+{
+ audio_io_handle_t output = AUDIO_IO_HANDLE_NONE;
+ uint32_t latency = 0;
+ status_t status;
+
+#ifdef AUDIO_POLICY_TEST
+ if (mCurOutput != 0) {
+ ALOGV("getOutput() test output mCurOutput %d, samplingRate %d, format %d, channelMask %x, mDirectOutput %d",
+ mCurOutput, mTestSamplingRate, mTestFormat, mTestChannels, mDirectOutput);
+
+ if (mTestOutputs[mCurOutput] == 0) {
+ ALOGV("getOutput() opening test output");
+ sp<AudioOutputDescriptor> outputDesc = new AudioOutputDescriptor(NULL);
+ outputDesc->mDevice = mTestDevice;
+ outputDesc->mLatency = mTestLatencyMs;
+ outputDesc->mFlags =
+ (audio_output_flags_t)(mDirectOutput ? AUDIO_OUTPUT_FLAG_DIRECT : 0);
+ outputDesc->mRefCount[stream] = 0;
+ audio_config_t config = AUDIO_CONFIG_INITIALIZER;
+ config.sample_rate = mTestSamplingRate;
+ config.channel_mask = mTestChannels;
+ config.format = mTestFormat;
+ if (offloadInfo != NULL) {
+ config.offload_info = *offloadInfo;
+ }
+ status = mpClientInterface->openOutput(0,
+ &mTestOutputs[mCurOutput],
+ &config,
+ &outputDesc->mDevice,
+ String8(""),
+ &outputDesc->mLatency,
+ outputDesc->mFlags);
+ if (status == NO_ERROR) {
+ outputDesc->mSamplingRate = config.sample_rate;
+ outputDesc->mFormat = config.format;
+ outputDesc->mChannelMask = config.channel_mask;
+ AudioParameter outputCmd = AudioParameter();
+ outputCmd.addInt(String8("set_id"),mCurOutput);
+ mpClientInterface->setParameters(mTestOutputs[mCurOutput],outputCmd.toString());
+ addOutput(mTestOutputs[mCurOutput], outputDesc);
+ }
+ }
+ return mTestOutputs[mCurOutput];
+ }
+#endif //AUDIO_POLICY_TEST
+
+ // open a direct output if required by specified parameters
+ //force direct flag if offload flag is set: offloading implies a direct output stream
+ // and all common behaviors are driven by checking only the direct flag
+ // this should normally be set appropriately in the policy configuration file
+ if ((flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) != 0) {
+ flags = (audio_output_flags_t)(flags | AUDIO_OUTPUT_FLAG_DIRECT);
+ }
+ if ((flags & AUDIO_OUTPUT_FLAG_HW_AV_SYNC) != 0) {
+ flags = (audio_output_flags_t)(flags | AUDIO_OUTPUT_FLAG_DIRECT);
+ }
+ // only allow deep buffering for music stream type
+ if (stream != AUDIO_STREAM_MUSIC) {
+ flags = (audio_output_flags_t)(flags &~AUDIO_OUTPUT_FLAG_DEEP_BUFFER);
+ }
+
+ sp<IOProfile> profile;
+
+ // skip direct output selection if the request can obviously be attached to a mixed output
+ // and not explicitly requested
+ if (((flags & AUDIO_OUTPUT_FLAG_DIRECT) == 0) &&
+ audio_is_linear_pcm(format) && samplingRate <= MAX_MIXER_SAMPLING_RATE &&
+ audio_channel_count_from_out_mask(channelMask) <= 2) {
+ goto non_direct_output;
+ }
+
+ // Do not allow offloading if one non offloadable effect is enabled. This prevents from
+ // creating an offloaded track and tearing it down immediately after start when audioflinger
+ // detects there is an active non offloadable effect.
+ // FIXME: We should check the audio session here but we do not have it in this context.
+ // This may prevent offloading in rare situations where effects are left active by apps
+ // in the background.
+
+ if (((flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) == 0) ||
+ !isNonOffloadableEffectEnabled()) {
+ profile = getProfileForDirectOutput(device,
+ samplingRate,
+ format,
+ channelMask,
+ (audio_output_flags_t)flags);
+ }
+
+ if (profile != 0) {
+ sp<AudioOutputDescriptor> outputDesc = NULL;
+
+ for (size_t i = 0; i < mOutputs.size(); i++) {
+ sp<AudioOutputDescriptor> desc = mOutputs.valueAt(i);
+ if (!desc->isDuplicated() && (profile == desc->mProfile)) {
+ outputDesc = desc;
+ // reuse direct output if currently open and configured with same parameters
+ if ((samplingRate == outputDesc->mSamplingRate) &&
+ (format == outputDesc->mFormat) &&
+ (channelMask == outputDesc->mChannelMask)) {
+ outputDesc->mDirectOpenCount++;
+ ALOGV("getOutput() reusing direct output %d", mOutputs.keyAt(i));
+ return mOutputs.keyAt(i);
+ }
+ }
+ }
+ // close direct output if currently open and configured with different parameters
+ if (outputDesc != NULL) {
+ closeOutput(outputDesc->mIoHandle);
+ }
+ outputDesc = new AudioOutputDescriptor(profile);
+ outputDesc->mDevice = device;
+ outputDesc->mLatency = 0;
+ outputDesc->mFlags =(audio_output_flags_t) (outputDesc->mFlags | flags);
+ audio_config_t config = AUDIO_CONFIG_INITIALIZER;
+ config.sample_rate = samplingRate;
+ config.channel_mask = channelMask;
+ config.format = format;
+ if (offloadInfo != NULL) {
+ config.offload_info = *offloadInfo;
+ }
+ status = mpClientInterface->openOutput(profile->mModule->mHandle,
+ &output,
+ &config,
+ &outputDesc->mDevice,
+ String8(""),
+ &outputDesc->mLatency,
+ outputDesc->mFlags);
+
+ // only accept an output with the requested parameters
+ if (status != NO_ERROR ||
+ (samplingRate != 0 && samplingRate != config.sample_rate) ||
+ (format != AUDIO_FORMAT_DEFAULT && format != config.format) ||
+ (channelMask != 0 && channelMask != config.channel_mask)) {
+ ALOGV("getOutput() failed opening direct output: output %d samplingRate %d %d,"
+ "format %d %d, channelMask %04x %04x", output, samplingRate,
+ outputDesc->mSamplingRate, format, outputDesc->mFormat, channelMask,
+ outputDesc->mChannelMask);
+ if (output != AUDIO_IO_HANDLE_NONE) {
+ mpClientInterface->closeOutput(output);
+ }
+ // fall back to mixer output if possible when the direct output could not be open
+ if (audio_is_linear_pcm(format) && samplingRate <= MAX_MIXER_SAMPLING_RATE) {
+ goto non_direct_output;
+ }
+ // fall back to mixer output if possible when the direct output could not be open
+ if (audio_is_linear_pcm(format) && samplingRate <= MAX_MIXER_SAMPLING_RATE) {
+ goto non_direct_output;
+ }
+ return AUDIO_IO_HANDLE_NONE;
+ }
+ outputDesc->mSamplingRate = config.sample_rate;
+ outputDesc->mChannelMask = config.channel_mask;
+ outputDesc->mFormat = config.format;
+ outputDesc->mRefCount[stream] = 0;
+ outputDesc->mStopTime[stream] = 0;
+ outputDesc->mDirectOpenCount = 1;
+
+ audio_io_handle_t srcOutput = getOutputForEffect();
+ addOutput(output, outputDesc);
+ audio_io_handle_t dstOutput = getOutputForEffect();
+ if (dstOutput == output) {
+ mpClientInterface->moveEffects(AUDIO_SESSION_OUTPUT_MIX, srcOutput, dstOutput);
+ }
+ mPreviousOutputs = mOutputs;
+ ALOGV("getOutput() returns new direct output %d", output);
+ mpClientInterface->onAudioPortListUpdate();
+ return output;
+ }
+
+non_direct_output:
+
+ // ignoring channel mask due to downmix capability in mixer
+
+ // open a non direct output
+
+ // for non direct outputs, only PCM is supported
+ if (audio_is_linear_pcm(format)) {
+ // get which output is suitable for the specified stream. The actual
+ // routing change will happen when startOutput() will be called
+ SortedVector<audio_io_handle_t> outputs = getOutputsForDevice(device, mOutputs);
+
+ // at this stage we should ignore the DIRECT flag as no direct output could be found earlier
+ flags = (audio_output_flags_t)(flags & ~AUDIO_OUTPUT_FLAG_DIRECT);
+ output = selectOutput(outputs, flags, format);
+ }
+ ALOGW_IF((output == 0), "getOutput() could not find output for stream %d, samplingRate %d,"
+ "format %d, channels %x, flags %x", stream, samplingRate, format, channelMask, flags);
+
+ ALOGV("getOutput() returns output %d", output);
+
+ return output;
+}
+
+audio_io_handle_t AudioPolicyManager::selectOutput(const SortedVector<audio_io_handle_t>& outputs,
+ audio_output_flags_t flags,
+ audio_format_t format)
+{
+ // select one output among several that provide a path to a particular device or set of
+ // devices (the list was previously build by getOutputsForDevice()).
+ // The priority is as follows:
+ // 1: the output with the highest number of requested policy flags
+ // 2: the primary output
+ // 3: the first output in the list
+
+ if (outputs.size() == 0) {
+ return 0;
+ }
+ if (outputs.size() == 1) {
+ return outputs[0];
+ }
+
+ int maxCommonFlags = 0;
+ audio_io_handle_t outputFlags = 0;
+ audio_io_handle_t outputPrimary = 0;
+
+ for (size_t i = 0; i < outputs.size(); i++) {
+ sp<AudioOutputDescriptor> outputDesc = mOutputs.valueFor(outputs[i]);
+ if (!outputDesc->isDuplicated()) {
+ // if a valid format is specified, skip output if not compatible
+ if (format != AUDIO_FORMAT_INVALID) {
+ if (outputDesc->mFlags & AUDIO_OUTPUT_FLAG_DIRECT) {
+ if (format != outputDesc->mFormat) {
+ continue;
+ }
+ } else if (!audio_is_linear_pcm(format)) {
+ continue;
+ }
+ }
+
+ int commonFlags = popcount(outputDesc->mProfile->mFlags & flags);
+ if (commonFlags > maxCommonFlags) {
+ outputFlags = outputs[i];
+ maxCommonFlags = commonFlags;
+ ALOGV("selectOutput() commonFlags for output %d, %04x", outputs[i], commonFlags);
+ }
+ if (outputDesc->mProfile->mFlags & AUDIO_OUTPUT_FLAG_PRIMARY) {
+ outputPrimary = outputs[i];
+ }
+ }
+ }
+
+ if (outputFlags != 0) {
+ return outputFlags;
+ }
+ if (outputPrimary != 0) {
+ return outputPrimary;
+ }
+
+ return outputs[0];
+}
+
+status_t AudioPolicyManager::startOutput(audio_io_handle_t output,
+ audio_stream_type_t stream,
+ audio_session_t session)
+{
+ ALOGV("startOutput() output %d, stream %d, session %d", output, stream, session);
+ ssize_t index = mOutputs.indexOfKey(output);
+ if (index < 0) {
+ ALOGW("startOutput() unknown output %d", output);
+ return BAD_VALUE;
+ }
+
+ // cannot start playback of STREAM_TTS if any other output is being used
+ uint32_t beaconMuteLatency = 0;
+ if (stream == AUDIO_STREAM_TTS) {
+ ALOGV("\t found BEACON stream");
+ if (isAnyOutputActive(AUDIO_STREAM_TTS /*streamToIgnore*/)) {
+ return INVALID_OPERATION;
+ } else {
+ beaconMuteLatency = handleEventForBeacon(STARTING_BEACON);
+ }
+ } else {
+ // some playback other than beacon starts
+ beaconMuteLatency = handleEventForBeacon(STARTING_OUTPUT);
+ }
+
+ sp<AudioOutputDescriptor> outputDesc = mOutputs.valueAt(index);
+
+ // increment usage count for this stream on the requested output:
+ // NOTE that the usage count is the same for duplicated output and hardware output which is
+ // necessary for a correct control of hardware output routing by startOutput() and stopOutput()
+ outputDesc->changeRefCount(stream, 1);
+
+ if (outputDesc->mRefCount[stream] == 1) {
+ // starting an output being rerouted?
+ audio_devices_t newDevice;
+ if (outputDesc->mPolicyMix != NULL) {
+ newDevice = AUDIO_DEVICE_OUT_REMOTE_SUBMIX;
+ } else {
+ newDevice = getNewOutputDevice(output, false /*fromCache*/);
+ }
+ routing_strategy strategy = getStrategy(stream);
+ bool shouldWait = (strategy == STRATEGY_SONIFICATION) ||
+ (strategy == STRATEGY_SONIFICATION_RESPECTFUL) ||
+ (beaconMuteLatency > 0);
+ uint32_t waitMs = beaconMuteLatency;
+ bool force = false;
+ for (size_t i = 0; i < mOutputs.size(); i++) {
+ sp<AudioOutputDescriptor> desc = mOutputs.valueAt(i);
+ if (desc != outputDesc) {
+ // force a device change if any other output is managed by the same hw
+ // module and has a current device selection that differs from selected device.
+ // In this case, the audio HAL must receive the new device selection so that it can
+ // change the device currently selected by the other active output.
+ if (outputDesc->sharesHwModuleWith(desc) &&
+ desc->device() != newDevice) {
+ force = true;
+ }
+ // wait for audio on other active outputs to be presented when starting
+ // a notification so that audio focus effect can propagate, or that a mute/unmute
+ // event occurred for beacon
+ uint32_t latency = desc->latency();
+ if (shouldWait && desc->isActive(latency * 2) && (waitMs < latency)) {
+ waitMs = latency;
+ }
+ }
+ }
+ uint32_t muteWaitMs = setOutputDevice(output, newDevice, force);
+
+ // handle special case for sonification while in call
+ if (isInCall()) {
+ handleIncallSonification(stream, true, false);
+ }
+
+ // apply volume rules for current stream and device if necessary
+ checkAndSetVolume(stream,
+ mStreams[stream].getVolumeIndex(newDevice),
+ output,
+ newDevice);
+
+ // update the outputs if starting an output with a stream that can affect notification
+ // routing
+ handleNotificationRoutingForStream(stream);
+
+ // Automatically enable the remote submix input when output is started on a re routing mix
+ // of type MIX_TYPE_RECORDERS
+ if (audio_is_remote_submix_device(newDevice) && outputDesc->mPolicyMix != NULL &&
+ outputDesc->mPolicyMix->mMixType == MIX_TYPE_RECORDERS) {
+ setDeviceConnectionStateInt(AUDIO_DEVICE_IN_REMOTE_SUBMIX,
+ AUDIO_POLICY_DEVICE_STATE_AVAILABLE,
+ outputDesc->mPolicyMix->mRegistrationId,
+ "remote-submix");
+ }
+
+ // force reevaluating accessibility routing when ringtone or alarm starts
+ if (strategy == STRATEGY_SONIFICATION) {
+ mpClientInterface->invalidateStream(AUDIO_STREAM_ACCESSIBILITY);
+ }
+
+ if (waitMs > muteWaitMs) {
+ usleep((waitMs - muteWaitMs) * 2 * 1000);
+ }
+ }
+ return NO_ERROR;
+}
+
+
+status_t AudioPolicyManager::stopOutput(audio_io_handle_t output,
+ audio_stream_type_t stream,
+ audio_session_t session)
+{
+ ALOGV("stopOutput() output %d, stream %d, session %d", output, stream, session);
+ ssize_t index = mOutputs.indexOfKey(output);
+ if (index < 0) {
+ ALOGW("stopOutput() unknown output %d", output);
+ return BAD_VALUE;
+ }
+
+ sp<AudioOutputDescriptor> outputDesc = mOutputs.valueAt(index);
+
+ // always handle stream stop, check which stream type is stopping
+ handleEventForBeacon(stream == AUDIO_STREAM_TTS ? STOPPING_BEACON : STOPPING_OUTPUT);
+
+ // handle special case for sonification while in call
+ if (isInCall()) {
+ handleIncallSonification(stream, false, false);
+ }
+
+ if (outputDesc->mRefCount[stream] > 0) {
+ // decrement usage count of this stream on the output
+ outputDesc->changeRefCount(stream, -1);
+ // store time at which the stream was stopped - see isStreamActive()
+ if (outputDesc->mRefCount[stream] == 0) {
+ // Automatically disable the remote submix input when output is stopped on a
+ // re routing mix of type MIX_TYPE_RECORDERS
+ if (audio_is_remote_submix_device(outputDesc->mDevice) &&
+ outputDesc->mPolicyMix != NULL &&
+ outputDesc->mPolicyMix->mMixType == MIX_TYPE_RECORDERS) {
+ setDeviceConnectionStateInt(AUDIO_DEVICE_IN_REMOTE_SUBMIX,
+ AUDIO_POLICY_DEVICE_STATE_UNAVAILABLE,
+ outputDesc->mPolicyMix->mRegistrationId,
+ "remote-submix");
+ }
+
+ outputDesc->mStopTime[stream] = systemTime();
+ audio_devices_t newDevice = getNewOutputDevice(output, false /*fromCache*/);
+ // delay the device switch by twice the latency because stopOutput() is executed when
+ // the track stop() command is received and at that time the audio track buffer can
+ // still contain data that needs to be drained. The latency only covers the audio HAL
+ // and kernel buffers. Also the latency does not always include additional delay in the
+ // audio path (audio DSP, CODEC ...)
+ setOutputDevice(output, newDevice, false, outputDesc->mLatency*2);
+
+ // force restoring the device selection on other active outputs if it differs from the
+ // one being selected for this output
+ for (size_t i = 0; i < mOutputs.size(); i++) {
+ audio_io_handle_t curOutput = mOutputs.keyAt(i);
+ sp<AudioOutputDescriptor> desc = mOutputs.valueAt(i);
+ if (curOutput != output &&
+ desc->isActive() &&
+ outputDesc->sharesHwModuleWith(desc) &&
+ (newDevice != desc->device())) {
+ setOutputDevice(curOutput,
+ getNewOutputDevice(curOutput, false /*fromCache*/),
+ true,
+ outputDesc->mLatency*2);
+ }
+ }
+ // update the outputs if stopping one with a stream that can affect notification routing
+ handleNotificationRoutingForStream(stream);
+ }
+ return NO_ERROR;
+ } else {
+ ALOGW("stopOutput() refcount is already 0 for output %d", output);
+ return INVALID_OPERATION;
+ }
+}
+
+void AudioPolicyManager::releaseOutput(audio_io_handle_t output,
+ audio_stream_type_t stream __unused,
+ audio_session_t session __unused)
+{
+ ALOGV("releaseOutput() %d", output);
+ ssize_t index = mOutputs.indexOfKey(output);
+ if (index < 0) {
+ ALOGW("releaseOutput() releasing unknown output %d", output);
+ return;
+ }
+
+#ifdef AUDIO_POLICY_TEST
+ int testIndex = testOutputIndex(output);
+ if (testIndex != 0) {
+ sp<AudioOutputDescriptor> outputDesc = mOutputs.valueAt(index);
+ if (outputDesc->isActive()) {
+ mpClientInterface->closeOutput(output);
+ mOutputs.removeItem(output);
+ mTestOutputs[testIndex] = 0;
+ }
+ return;
+ }
+#endif //AUDIO_POLICY_TEST
+
+ sp<AudioOutputDescriptor> desc = mOutputs.valueAt(index);
+ if (desc->mFlags & AUDIO_OUTPUT_FLAG_DIRECT) {
+ if (desc->mDirectOpenCount <= 0) {
+ ALOGW("releaseOutput() invalid open count %d for output %d",
+ desc->mDirectOpenCount, output);
+ return;
+ }
+ if (--desc->mDirectOpenCount == 0) {
+ closeOutput(output);
+ // If effects where present on the output, audioflinger moved them to the primary
+ // output by default: move them back to the appropriate output.
+ audio_io_handle_t dstOutput = getOutputForEffect();
+ if (dstOutput != mPrimaryOutput) {
+ mpClientInterface->moveEffects(AUDIO_SESSION_OUTPUT_MIX, mPrimaryOutput, dstOutput);
+ }
+ mpClientInterface->onAudioPortListUpdate();
+ }
+ }
+}
+
+
+status_t AudioPolicyManager::getInputForAttr(const audio_attributes_t *attr,
+ audio_io_handle_t *input,
+ audio_session_t session,
+ uint32_t samplingRate,
+ audio_format_t format,
+ audio_channel_mask_t channelMask,
+ audio_input_flags_t flags,
+ input_type_t *inputType)
+{
+ ALOGV("getInputForAttr() source %d, samplingRate %d, format %d, channelMask %x,"
+ "session %d, flags %#x",
+ attr->source, samplingRate, format, channelMask, session, flags);
+
+ *input = AUDIO_IO_HANDLE_NONE;
+ *inputType = API_INPUT_INVALID;
+ audio_devices_t device;
+ // handle legacy remote submix case where the address was not always specified
+ String8 address = String8("");
+ bool isSoundTrigger = false;
+ audio_source_t inputSource = attr->source;
+ audio_source_t halInputSource;
+ AudioMix *policyMix = NULL;
+
+ if (inputSource == AUDIO_SOURCE_DEFAULT) {
+ inputSource = AUDIO_SOURCE_MIC;
+ }
+ halInputSource = inputSource;
+
+ if (inputSource == AUDIO_SOURCE_REMOTE_SUBMIX &&
+ strncmp(attr->tags, "addr=", strlen("addr=")) == 0) {
+ device = AUDIO_DEVICE_IN_REMOTE_SUBMIX;
+ address = String8(attr->tags + strlen("addr="));
+ ssize_t index = mPolicyMixes.indexOfKey(address);
+ if (index < 0) {
+ ALOGW("getInputForAttr() no policy for address %s", address.string());
+ return BAD_VALUE;
+ }
+ if (mPolicyMixes[index]->mMix.mMixType != MIX_TYPE_PLAYERS) {
+ ALOGW("getInputForAttr() bad policy mix type for address %s", address.string());
+ return BAD_VALUE;
+ }
+ policyMix = &mPolicyMixes[index]->mMix;
+ *inputType = API_INPUT_MIX_EXT_POLICY_REROUTE;
+ } else {
+ device = getDeviceAndMixForInputSource(inputSource, &policyMix);
+ if (device == AUDIO_DEVICE_NONE) {
+ ALOGW("getInputForAttr() could not find device for source %d", inputSource);
+ return BAD_VALUE;
+ }
+ if (policyMix != NULL) {
+ address = policyMix->mRegistrationId;
+ if (policyMix->mMixType == MIX_TYPE_RECORDERS) {
+ // there is an external policy, but this input is attached to a mix of recorders,
+ // meaning it receives audio injected into the framework, so the recorder doesn't
+ // know about it and is therefore considered "legacy"
+ *inputType = API_INPUT_LEGACY;
+ } else {
+ // recording a mix of players defined by an external policy, we're rerouting for
+ // an external policy
+ *inputType = API_INPUT_MIX_EXT_POLICY_REROUTE;
+ }
+ } else if (audio_is_remote_submix_device(device)) {
+ address = String8("0");
+ *inputType = API_INPUT_MIX_CAPTURE;
+ } else {
+ *inputType = API_INPUT_LEGACY;
+ }
+ // adapt channel selection to input source
+ switch (inputSource) {
+ case AUDIO_SOURCE_VOICE_UPLINK:
+ channelMask = AUDIO_CHANNEL_IN_VOICE_UPLINK;
+ break;
+ case AUDIO_SOURCE_VOICE_DOWNLINK:
+ channelMask = AUDIO_CHANNEL_IN_VOICE_DNLINK;
+ break;
+ case AUDIO_SOURCE_VOICE_CALL:
+ channelMask = AUDIO_CHANNEL_IN_VOICE_UPLINK | AUDIO_CHANNEL_IN_VOICE_DNLINK;
+ break;
+ default:
+ break;
+ }
+ if (inputSource == AUDIO_SOURCE_HOTWORD) {
+ ssize_t index = mSoundTriggerSessions.indexOfKey(session);
+ if (index >= 0) {
+ *input = mSoundTriggerSessions.valueFor(session);
+ isSoundTrigger = true;
+ flags = (audio_input_flags_t)(flags | AUDIO_INPUT_FLAG_HW_HOTWORD);
+ ALOGV("SoundTrigger capture on session %d input %d", session, *input);
+ } else {
+ halInputSource = AUDIO_SOURCE_VOICE_RECOGNITION;
+ }
+ }
+ }
+
+ sp<IOProfile> profile = getInputProfile(device, address,
+ samplingRate, format, channelMask,
+ flags);
+ if (profile == 0) {
+ //retry without flags
+ audio_input_flags_t log_flags = flags;
+ flags = AUDIO_INPUT_FLAG_NONE;
+ profile = getInputProfile(device, address,
+ samplingRate, format, channelMask,
+ flags);
+ if (profile == 0) {
+ ALOGW("getInputForAttr() could not find profile for device 0x%X, samplingRate %u,"
+ "format %#x, channelMask 0x%X, flags %#x",
+ device, samplingRate, format, channelMask, log_flags);
+ return BAD_VALUE;
+ }
+ }
+
+ if (profile->mModule->mHandle == 0) {
+ ALOGE("getInputForAttr(): HW module %s not opened", profile->mModule->mName);
+ return NO_INIT;
+ }
+
+ audio_config_t config = AUDIO_CONFIG_INITIALIZER;
+ config.sample_rate = samplingRate;
+ config.channel_mask = channelMask;
+ config.format = format;
+
+ status_t status = mpClientInterface->openInput(profile->mModule->mHandle,
+ input,
+ &config,
+ &device,
+ address,
+ halInputSource,
+ flags);
+
+ // only accept input with the exact requested set of parameters
+ if (status != NO_ERROR || *input == AUDIO_IO_HANDLE_NONE ||
+ (samplingRate != config.sample_rate) ||
+ (format != config.format) ||
+ (channelMask != config.channel_mask)) {
+ ALOGW("getInputForAttr() failed opening input: samplingRate %d, format %d, channelMask %x",
+ samplingRate, format, channelMask);
+ if (*input != AUDIO_IO_HANDLE_NONE) {
+ mpClientInterface->closeInput(*input);
+ }
+ return BAD_VALUE;
+ }
+
+ sp<AudioInputDescriptor> inputDesc = new AudioInputDescriptor(profile);
+ inputDesc->mInputSource = inputSource;
+ inputDesc->mRefCount = 0;
+ inputDesc->mOpenRefCount = 1;
+ inputDesc->mSamplingRate = samplingRate;
+ inputDesc->mFormat = format;
+ inputDesc->mChannelMask = channelMask;
+ inputDesc->mDevice = device;
+ inputDesc->mSessions.add(session);
+ inputDesc->mIsSoundTrigger = isSoundTrigger;
+ inputDesc->mPolicyMix = policyMix;
+
+ ALOGV("getInputForAttr() returns input type = %d", inputType);
+
+ addInput(*input, inputDesc);
+ mpClientInterface->onAudioPortListUpdate();
+ return NO_ERROR;
+}
+
+status_t AudioPolicyManager::startInput(audio_io_handle_t input,
+ audio_session_t session)
+{
+ ALOGV("startInput() input %d", input);
+ ssize_t index = mInputs.indexOfKey(input);
+ if (index < 0) {
+ ALOGW("startInput() unknown input %d", input);
+ return BAD_VALUE;
+ }
+ sp<AudioInputDescriptor> inputDesc = mInputs.valueAt(index);
+
+ index = inputDesc->mSessions.indexOf(session);
+ if (index < 0) {
+ ALOGW("startInput() unknown session %d on input %d", session, input);
+ return BAD_VALUE;
+ }
+
+ // virtual input devices are compatible with other input devices
+ if (!isVirtualInputDevice(inputDesc->mDevice)) {
+
+ // for a non-virtual input device, check if there is another (non-virtual) active input
+ audio_io_handle_t activeInput = getActiveInput();
+ if (activeInput != 0 && activeInput != input) {
+
+ // If the already active input uses AUDIO_SOURCE_HOTWORD then it is closed,
+ // otherwise the active input continues and the new input cannot be started.
+ sp<AudioInputDescriptor> activeDesc = mInputs.valueFor(activeInput);
+ if (activeDesc->mInputSource == AUDIO_SOURCE_HOTWORD) {
+ ALOGW("startInput(%d) preempting low-priority input %d", input, activeInput);
+ stopInput(activeInput, activeDesc->mSessions.itemAt(0));
+ releaseInput(activeInput, activeDesc->mSessions.itemAt(0));
+ } else {
+ ALOGE("startInput(%d) failed: other input %d already started", input, activeInput);
+ return INVALID_OPERATION;
+ }
+ }
+ }
+
+ if (inputDesc->mRefCount == 0) {
+ if (activeInputsCount() == 0) {
+ SoundTrigger::setCaptureState(true);
+ }
+ setInputDevice(input, getNewInputDevice(input), true /* force */);
+
+ // automatically enable the remote submix output when input is started if not
+ // used by a policy mix of type MIX_TYPE_RECORDERS
+ // For remote submix (a virtual device), we open only one input per capture request.
+ if (audio_is_remote_submix_device(inputDesc->mDevice)) {
+ String8 address = String8("");
+ if (inputDesc->mPolicyMix == NULL) {
+ address = String8("0");
+ } else if (inputDesc->mPolicyMix->mMixType == MIX_TYPE_PLAYERS) {
+ address = inputDesc->mPolicyMix->mRegistrationId;
+ }
+ if (address != "") {
+ setDeviceConnectionStateInt(AUDIO_DEVICE_OUT_REMOTE_SUBMIX,
+ AUDIO_POLICY_DEVICE_STATE_AVAILABLE,
+ address, "remote-submix");
+ }
+ }
+ }
+
+ ALOGV("AudioPolicyManager::startInput() input source = %d", inputDesc->mInputSource);
+
+ inputDesc->mRefCount++;
+ return NO_ERROR;
+}
+
+status_t AudioPolicyManager::stopInput(audio_io_handle_t input,
+ audio_session_t session)
+{
+ ALOGV("stopInput() input %d", input);
+ ssize_t index = mInputs.indexOfKey(input);
+ if (index < 0) {
+ ALOGW("stopInput() unknown input %d", input);
+ return BAD_VALUE;
+ }
+ sp<AudioInputDescriptor> inputDesc = mInputs.valueAt(index);
+
+ index = inputDesc->mSessions.indexOf(session);
+ if (index < 0) {
+ ALOGW("stopInput() unknown session %d on input %d", session, input);
+ return BAD_VALUE;
+ }
+
+ if (inputDesc->mRefCount == 0) {
+ ALOGW("stopInput() input %d already stopped", input);
+ return INVALID_OPERATION;
+ }
+
+ inputDesc->mRefCount--;
+ if (inputDesc->mRefCount == 0) {
+
+ // automatically disable the remote submix output when input is stopped if not
+ // used by a policy mix of type MIX_TYPE_RECORDERS
+ if (audio_is_remote_submix_device(inputDesc->mDevice)) {
+ String8 address = String8("");
+ if (inputDesc->mPolicyMix == NULL) {
+ address = String8("0");
+ } else if (inputDesc->mPolicyMix->mMixType == MIX_TYPE_PLAYERS) {
+ address = inputDesc->mPolicyMix->mRegistrationId;
+ }
+ if (address != "") {
+ setDeviceConnectionStateInt(AUDIO_DEVICE_OUT_REMOTE_SUBMIX,
+ AUDIO_POLICY_DEVICE_STATE_UNAVAILABLE,
+ address, "remote-submix");
+ }
+ }
+
+ resetInputDevice(input);
+
+ if (activeInputsCount() == 0) {
+ SoundTrigger::setCaptureState(false);
+ }
+ }
+ return NO_ERROR;
+}
+
+void AudioPolicyManager::releaseInput(audio_io_handle_t input,
+ audio_session_t session)
+{
+ ALOGV("releaseInput() %d", input);
+ ssize_t index = mInputs.indexOfKey(input);
+ if (index < 0) {
+ ALOGW("releaseInput() releasing unknown input %d", input);
+ return;
+ }
+ sp<AudioInputDescriptor> inputDesc = mInputs.valueAt(index);
+ ALOG_ASSERT(inputDesc != 0);
+
+ index = inputDesc->mSessions.indexOf(session);
+ if (index < 0) {
+ ALOGW("releaseInput() unknown session %d on input %d", session, input);
+ return;
+ }
+ inputDesc->mSessions.remove(session);
+ if (inputDesc->mOpenRefCount == 0) {
+ ALOGW("releaseInput() invalid open ref count %d", inputDesc->mOpenRefCount);
+ return;
+ }
+ inputDesc->mOpenRefCount--;
+ if (inputDesc->mOpenRefCount > 0) {
+ ALOGV("releaseInput() exit > 0");
+ return;
+ }
+
+ closeInput(input);
+ mpClientInterface->onAudioPortListUpdate();
+ ALOGV("releaseInput() exit");
+}
+
+void AudioPolicyManager::closeAllInputs() {
+ bool patchRemoved = false;
+
+ for(size_t input_index = 0; input_index < mInputs.size(); input_index++) {
+ sp<AudioInputDescriptor> inputDesc = mInputs.valueAt(input_index);
+ ssize_t patch_index = mAudioPatches.indexOfKey(inputDesc->mPatchHandle);
+ if (patch_index >= 0) {
+ sp<AudioPatch> patchDesc = mAudioPatches.valueAt(patch_index);
+ status_t status = mpClientInterface->releaseAudioPatch(patchDesc->mAfPatchHandle, 0);
+ mAudioPatches.removeItemsAt(patch_index);
+ patchRemoved = true;
+ }
+ mpClientInterface->closeInput(mInputs.keyAt(input_index));
+ }
+ mInputs.clear();
+ nextAudioPortGeneration();
+
+ if (patchRemoved) {
+ mpClientInterface->onAudioPatchListUpdate();
+ }
+}
+
+void AudioPolicyManager::initStreamVolume(audio_stream_type_t stream,
+ int indexMin,
+ int indexMax)
+{
+ ALOGV("initStreamVolume() stream %d, min %d, max %d", stream , indexMin, indexMax);
+ if (indexMin < 0 || indexMin >= indexMax) {
+ ALOGW("initStreamVolume() invalid index limits for stream %d, min %d, max %d", stream , indexMin, indexMax);
+ return;
+ }
+ mStreams[stream].mIndexMin = indexMin;
+ mStreams[stream].mIndexMax = indexMax;
+ //FIXME: AUDIO_STREAM_ACCESSIBILITY volume follows AUDIO_STREAM_MUSIC for now
+ if (stream == AUDIO_STREAM_MUSIC) {
+ mStreams[AUDIO_STREAM_ACCESSIBILITY].mIndexMin = indexMin;
+ mStreams[AUDIO_STREAM_ACCESSIBILITY].mIndexMax = indexMax;
+ }
+}
+
+status_t AudioPolicyManager::setStreamVolumeIndex(audio_stream_type_t stream,
+ int index,
+ audio_devices_t device)
+{
+
+ if ((index < mStreams[stream].mIndexMin) || (index > mStreams[stream].mIndexMax)) {
+ return BAD_VALUE;
+ }
+ if (!audio_is_output_device(device)) {
+ return BAD_VALUE;
+ }
+
+ // Force max volume if stream cannot be muted
+ if (!mStreams[stream].mCanBeMuted) index = mStreams[stream].mIndexMax;
+
+ ALOGV("setStreamVolumeIndex() stream %d, device %04x, index %d",
+ stream, device, index);
+
+ // if device is AUDIO_DEVICE_OUT_DEFAULT set default value and
+ // clear all device specific values
+ if (device == AUDIO_DEVICE_OUT_DEFAULT) {
+ mStreams[stream].mIndexCur.clear();
+ }
+ mStreams[stream].mIndexCur.add(device, index);
+
+ // update volume on all outputs whose current device is also selected by the same
+ // strategy as the device specified by the caller
+ audio_devices_t strategyDevice = getDeviceForStrategy(getStrategy(stream), true /*fromCache*/);
+
+
+ //FIXME: AUDIO_STREAM_ACCESSIBILITY volume follows AUDIO_STREAM_MUSIC for now
+ audio_devices_t accessibilityDevice = AUDIO_DEVICE_NONE;
+ if (stream == AUDIO_STREAM_MUSIC) {
+ mStreams[AUDIO_STREAM_ACCESSIBILITY].mIndexCur.add(device, index);
+ accessibilityDevice = getDeviceForStrategy(STRATEGY_ACCESSIBILITY, true /*fromCache*/);
+ }
+ if ((device != AUDIO_DEVICE_OUT_DEFAULT) &&
+ (device & (strategyDevice | accessibilityDevice)) == 0) {
+ return NO_ERROR;
+ }
+ status_t status = NO_ERROR;
+ for (size_t i = 0; i < mOutputs.size(); i++) {
+ audio_devices_t curDevice =
+ ApmGains::getDeviceForVolume(mOutputs.valueAt(i)->device());
+ if ((device == AUDIO_DEVICE_OUT_DEFAULT) || ((curDevice & strategyDevice) != 0)) {
+ status_t volStatus = checkAndSetVolume(stream, index, mOutputs.keyAt(i), curDevice);
+ if (volStatus != NO_ERROR) {
+ status = volStatus;
+ }
+ }
+ if ((device == AUDIO_DEVICE_OUT_DEFAULT) || ((curDevice & accessibilityDevice) != 0)) {
+ status_t volStatus = checkAndSetVolume(AUDIO_STREAM_ACCESSIBILITY,
+ index, mOutputs.keyAt(i), curDevice);
+ }
+ }
+ return status;
+}
+
+status_t AudioPolicyManager::getStreamVolumeIndex(audio_stream_type_t stream,
+ int *index,
+ audio_devices_t device)
+{
+ if (index == NULL) {
+ return BAD_VALUE;
+ }
+ if (!audio_is_output_device(device)) {
+ return BAD_VALUE;
+ }
+ // if device is AUDIO_DEVICE_OUT_DEFAULT, return volume for device corresponding to
+ // the strategy the stream belongs to.
+ if (device == AUDIO_DEVICE_OUT_DEFAULT) {
+ device = getDeviceForStrategy(getStrategy(stream), true /*fromCache*/);
+ }
+ device = ApmGains::getDeviceForVolume(device);
+
+ *index = mStreams[stream].getVolumeIndex(device);
+ ALOGV("getStreamVolumeIndex() stream %d device %08x index %d", stream, device, *index);
+ return NO_ERROR;
+}
+
+audio_io_handle_t AudioPolicyManager::selectOutputForEffects(
+ const SortedVector<audio_io_handle_t>& outputs)
+{
+ // select one output among several suitable for global effects.
+ // The priority is as follows:
+ // 1: An offloaded output. If the effect ends up not being offloadable,
+ // AudioFlinger will invalidate the track and the offloaded output
+ // will be closed causing the effect to be moved to a PCM output.
+ // 2: A deep buffer output
+ // 3: the first output in the list
+
+ if (outputs.size() == 0) {
+ return 0;
+ }
+
+ audio_io_handle_t outputOffloaded = 0;
+ audio_io_handle_t outputDeepBuffer = 0;
+
+ for (size_t i = 0; i < outputs.size(); i++) {
+ sp<AudioOutputDescriptor> desc = mOutputs.valueFor(outputs[i]);
+ ALOGV("selectOutputForEffects outputs[%zu] flags %x", i, desc->mFlags);
+ if ((desc->mFlags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) != 0) {
+ outputOffloaded = outputs[i];
+ }
+ if ((desc->mFlags & AUDIO_OUTPUT_FLAG_DEEP_BUFFER) != 0) {
+ outputDeepBuffer = outputs[i];
+ }
+ }
+
+ ALOGV("selectOutputForEffects outputOffloaded %d outputDeepBuffer %d",
+ outputOffloaded, outputDeepBuffer);
+ if (outputOffloaded != 0) {
+ return outputOffloaded;
+ }
+ if (outputDeepBuffer != 0) {
+ return outputDeepBuffer;
+ }
+
+ return outputs[0];
+}
+
+audio_io_handle_t AudioPolicyManager::getOutputForEffect(const effect_descriptor_t *desc)
+{
+ // apply simple rule where global effects are attached to the same output as MUSIC streams
+
+ routing_strategy strategy = getStrategy(AUDIO_STREAM_MUSIC);
+ audio_devices_t device = getDeviceForStrategy(strategy, false /*fromCache*/);
+ SortedVector<audio_io_handle_t> dstOutputs = getOutputsForDevice(device, mOutputs);
+
+ audio_io_handle_t output = selectOutputForEffects(dstOutputs);
+ ALOGV("getOutputForEffect() got output %d for fx %s flags %x",
+ output, (desc == NULL) ? "unspecified" : desc->name, (desc == NULL) ? 0 : desc->flags);
+
+ return output;
+}
+
+status_t AudioPolicyManager::registerEffect(const effect_descriptor_t *desc,
+ audio_io_handle_t io,
+ uint32_t strategy,
+ int session,
+ int id)
+{
+ ssize_t index = mOutputs.indexOfKey(io);
+ if (index < 0) {
+ index = mInputs.indexOfKey(io);
+ if (index < 0) {
+ ALOGW("registerEffect() unknown io %d", io);
+ return INVALID_OPERATION;
+ }
+ }
+
+ if (mTotalEffectsMemory + desc->memoryUsage > getMaxEffectsMemory()) {
+ ALOGW("registerEffect() memory limit exceeded for Fx %s, Memory %d KB",
+ desc->name, desc->memoryUsage);
+ return INVALID_OPERATION;
+ }
+ mTotalEffectsMemory += desc->memoryUsage;
+ ALOGV("registerEffect() effect %s, io %d, strategy %d session %d id %d",
+ desc->name, io, strategy, session, id);
+ ALOGV("registerEffect() memory %d, total memory %d", desc->memoryUsage, mTotalEffectsMemory);
+
+ sp<EffectDescriptor> effectDesc = new EffectDescriptor();
+ memcpy (&effectDesc->mDesc, desc, sizeof(effect_descriptor_t));
+ effectDesc->mIo = io;
+ effectDesc->mStrategy = (routing_strategy)strategy;
+ effectDesc->mSession = session;
+ effectDesc->mEnabled = false;
+
+ mEffects.add(id, effectDesc);
+
+ return NO_ERROR;
+}
+
+status_t AudioPolicyManager::unregisterEffect(int id)
+{
+ ssize_t index = mEffects.indexOfKey(id);
+ if (index < 0) {
+ ALOGW("unregisterEffect() unknown effect ID %d", id);
+ return INVALID_OPERATION;
+ }
+
+ sp<EffectDescriptor> effectDesc = mEffects.valueAt(index);
+
+ setEffectEnabled(effectDesc, false);
+
+ if (mTotalEffectsMemory < effectDesc->mDesc.memoryUsage) {
+ ALOGW("unregisterEffect() memory %d too big for total %d",
+ effectDesc->mDesc.memoryUsage, mTotalEffectsMemory);
+ effectDesc->mDesc.memoryUsage = mTotalEffectsMemory;
+ }
+ mTotalEffectsMemory -= effectDesc->mDesc.memoryUsage;
+ ALOGV("unregisterEffect() effect %s, ID %d, memory %d total memory %d",
+ effectDesc->mDesc.name, id, effectDesc->mDesc.memoryUsage, mTotalEffectsMemory);
+
+ mEffects.removeItem(id);
+
+ return NO_ERROR;
+}
+
+status_t AudioPolicyManager::setEffectEnabled(int id, bool enabled)
+{
+ ssize_t index = mEffects.indexOfKey(id);
+ if (index < 0) {
+ ALOGW("unregisterEffect() unknown effect ID %d", id);
+ return INVALID_OPERATION;
+ }
+
+ return setEffectEnabled(mEffects.valueAt(index), enabled);
+}
+
+status_t AudioPolicyManager::setEffectEnabled(const sp<EffectDescriptor>& effectDesc, bool enabled)
+{
+ if (enabled == effectDesc->mEnabled) {
+ ALOGV("setEffectEnabled(%s) effect already %s",
+ enabled?"true":"false", enabled?"enabled":"disabled");
+ return INVALID_OPERATION;
+ }
+
+ if (enabled) {
+ if (mTotalEffectsCpuLoad + effectDesc->mDesc.cpuLoad > getMaxEffectsCpuLoad()) {
+ ALOGW("setEffectEnabled(true) CPU Load limit exceeded for Fx %s, CPU %f MIPS",
+ effectDesc->mDesc.name, (float)effectDesc->mDesc.cpuLoad/10);
+ return INVALID_OPERATION;
+ }
+ mTotalEffectsCpuLoad += effectDesc->mDesc.cpuLoad;
+ ALOGV("setEffectEnabled(true) total CPU %d", mTotalEffectsCpuLoad);
+ } else {
+ if (mTotalEffectsCpuLoad < effectDesc->mDesc.cpuLoad) {
+ ALOGW("setEffectEnabled(false) CPU load %d too high for total %d",
+ effectDesc->mDesc.cpuLoad, mTotalEffectsCpuLoad);
+ effectDesc->mDesc.cpuLoad = mTotalEffectsCpuLoad;
+ }
+ mTotalEffectsCpuLoad -= effectDesc->mDesc.cpuLoad;
+ ALOGV("setEffectEnabled(false) total CPU %d", mTotalEffectsCpuLoad);
+ }
+ effectDesc->mEnabled = enabled;
+ return NO_ERROR;
+}
+
+bool AudioPolicyManager::isNonOffloadableEffectEnabled()
+{
+ for (size_t i = 0; i < mEffects.size(); i++) {
+ sp<EffectDescriptor> effectDesc = mEffects.valueAt(i);
+ if (effectDesc->mEnabled && (effectDesc->mStrategy == STRATEGY_MEDIA) &&
+ ((effectDesc->mDesc.flags & EFFECT_FLAG_OFFLOAD_SUPPORTED) == 0)) {
+ ALOGV("isNonOffloadableEffectEnabled() non offloadable effect %s enabled on session %d",
+ effectDesc->mDesc.name, effectDesc->mSession);
+ return true;
+ }
+ }
+ return false;
+}
+
+bool AudioPolicyManager::isStreamActive(audio_stream_type_t stream, uint32_t inPastMs) const
+{
+ nsecs_t sysTime = systemTime();
+ for (size_t i = 0; i < mOutputs.size(); i++) {
+ const sp<AudioOutputDescriptor> outputDesc = mOutputs.valueAt(i);
+ if (outputDesc->isStreamActive(stream, inPastMs, sysTime)) {
+ return true;
+ }
+ }
+ return false;
+}
+
+bool AudioPolicyManager::isStreamActiveRemotely(audio_stream_type_t stream,
+ uint32_t inPastMs) const
+{
+ nsecs_t sysTime = systemTime();
+ for (size_t i = 0; i < mOutputs.size(); i++) {
+ const sp<AudioOutputDescriptor> outputDesc = mOutputs.valueAt(i);
+ if (((outputDesc->device() & APM_AUDIO_OUT_DEVICE_REMOTE_ALL) != 0) &&
+ outputDesc->isStreamActive(stream, inPastMs, sysTime)) {
+ // do not consider re routing (when the output is going to a dynamic policy)
+ // as "remote playback"
+ if (outputDesc->mPolicyMix == NULL) {
+ return true;
+ }
+ }
+ }
+ return false;
+}
+
+bool AudioPolicyManager::isSourceActive(audio_source_t source) const
+{
+ for (size_t i = 0; i < mInputs.size(); i++) {
+ const sp<AudioInputDescriptor> inputDescriptor = mInputs.valueAt(i);
+ if (inputDescriptor->mRefCount == 0) {
+ continue;
+ }
+ if (inputDescriptor->mInputSource == (int)source) {
+ return true;
+ }
+ // AUDIO_SOURCE_HOTWORD is equivalent to AUDIO_SOURCE_VOICE_RECOGNITION only if it
+ // corresponds to an active capture triggered by a hardware hotword recognition
+ if ((source == AUDIO_SOURCE_VOICE_RECOGNITION) &&
+ (inputDescriptor->mInputSource == AUDIO_SOURCE_HOTWORD)) {
+ // FIXME: we should not assume that the first session is the active one and keep
+ // activity count per session. Same in startInput().
+ ssize_t index = mSoundTriggerSessions.indexOfKey(inputDescriptor->mSessions.itemAt(0));
+ if (index >= 0) {
+ return true;
+ }
+ }
+ }
+ return false;
+}
+
+// Register a list of custom mixes with their attributes and format.
+// When a mix is registered, corresponding input and output profiles are
+// added to the remote submix hw module. The profile contains only the
+// parameters (sampling rate, format...) specified by the mix.
+// The corresponding input remote submix device is also connected.
+//
+// When a remote submix device is connected, the address is checked to select the
+// appropriate profile and the corresponding input or output stream is opened.
+//
+// When capture starts, getInputForAttr() will:
+// - 1 look for a mix matching the address passed in attribtutes tags if any
+// - 2 if none found, getDeviceForInputSource() will:
+// - 2.1 look for a mix matching the attributes source
+// - 2.2 if none found, default to device selection by policy rules
+// At this time, the corresponding output remote submix device is also connected
+// and active playback use cases can be transferred to this mix if needed when reconnecting
+// after AudioTracks are invalidated
+//
+// When playback starts, getOutputForAttr() will:
+// - 1 look for a mix matching the address passed in attribtutes tags if any
+// - 2 if none found, look for a mix matching the attributes usage
+// - 3 if none found, default to device and output selection by policy rules.
+
+status_t AudioPolicyManager::registerPolicyMixes(Vector<AudioMix> mixes)
+{
+ sp<HwModule> module;
+ for (size_t i = 0; i < mHwModules.size(); i++) {
+ if (strcmp(AUDIO_HARDWARE_MODULE_ID_REMOTE_SUBMIX, mHwModules[i]->mName) == 0 &&
+ mHwModules[i]->mHandle != 0) {
+ module = mHwModules[i];
+ break;
+ }
+ }
+
+ if (module == 0) {
+ return INVALID_OPERATION;
+ }
+
+ ALOGV("registerPolicyMixes() num mixes %d", mixes.size());
+
+ for (size_t i = 0; i < mixes.size(); i++) {
+ String8 address = mixes[i].mRegistrationId;
+ ssize_t index = mPolicyMixes.indexOfKey(address);
+ if (index >= 0) {
+ ALOGE("registerPolicyMixes(): mix for address %s already registered", address.string());
+ continue;
+ }
+ audio_config_t outputConfig = mixes[i].mFormat;
+ audio_config_t inputConfig = mixes[i].mFormat;
+ // NOTE: audio flinger mixer does not support mono output: configure remote submix HAL in
+ // stereo and let audio flinger do the channel conversion if needed.
+ outputConfig.channel_mask = AUDIO_CHANNEL_OUT_STEREO;
+ inputConfig.channel_mask = AUDIO_CHANNEL_IN_STEREO;
+ module->addOutputProfile(address, &outputConfig,
+ AUDIO_DEVICE_OUT_REMOTE_SUBMIX, address);
+ module->addInputProfile(address, &inputConfig,
+ AUDIO_DEVICE_IN_REMOTE_SUBMIX, address);
+ sp<AudioPolicyMix> policyMix = new AudioPolicyMix();
+ policyMix->mMix = mixes[i];
+ mPolicyMixes.add(address, policyMix);
+ if (mixes[i].mMixType == MIX_TYPE_PLAYERS) {
+ setDeviceConnectionStateInt(AUDIO_DEVICE_IN_REMOTE_SUBMIX,
+ AUDIO_POLICY_DEVICE_STATE_AVAILABLE,
+ address.string(), "remote-submix");
+ } else {
+ setDeviceConnectionStateInt(AUDIO_DEVICE_OUT_REMOTE_SUBMIX,
+ AUDIO_POLICY_DEVICE_STATE_AVAILABLE,
+ address.string(), "remote-submix");
+ }
+ }
+ return NO_ERROR;
+}
+
+status_t AudioPolicyManager::unregisterPolicyMixes(Vector<AudioMix> mixes)
+{
+ sp<HwModule> module;
+ for (size_t i = 0; i < mHwModules.size(); i++) {
+ if (strcmp(AUDIO_HARDWARE_MODULE_ID_REMOTE_SUBMIX, mHwModules[i]->mName) == 0 &&
+ mHwModules[i]->mHandle != 0) {
+ module = mHwModules[i];
+ break;
+ }
+ }
+
+ if (module == 0) {
+ return INVALID_OPERATION;
+ }
+
+ ALOGV("unregisterPolicyMixes() num mixes %d", mixes.size());
+
+ for (size_t i = 0; i < mixes.size(); i++) {
+ String8 address = mixes[i].mRegistrationId;
+ ssize_t index = mPolicyMixes.indexOfKey(address);
+ if (index < 0) {
+ ALOGE("unregisterPolicyMixes(): mix for address %s not registered", address.string());
+ continue;
+ }
+
+ mPolicyMixes.removeItemsAt(index);
+
+ if (getDeviceConnectionState(AUDIO_DEVICE_IN_REMOTE_SUBMIX, address.string()) ==
+ AUDIO_POLICY_DEVICE_STATE_AVAILABLE)
+ {
+ setDeviceConnectionStateInt(AUDIO_DEVICE_IN_REMOTE_SUBMIX,
+ AUDIO_POLICY_DEVICE_STATE_UNAVAILABLE,
+ address.string(), "remote-submix");
+ }
+
+ if (getDeviceConnectionState(AUDIO_DEVICE_OUT_REMOTE_SUBMIX, address.string()) ==
+ AUDIO_POLICY_DEVICE_STATE_AVAILABLE)
+ {
+ setDeviceConnectionStateInt(AUDIO_DEVICE_OUT_REMOTE_SUBMIX,
+ AUDIO_POLICY_DEVICE_STATE_UNAVAILABLE,
+ address.string(), "remote-submix");
+ }
+ module->removeOutputProfile(address);
+ module->removeInputProfile(address);
+ }
+ return NO_ERROR;
+}
+
+
+status_t AudioPolicyManager::dump(int fd)
+{
+ const size_t SIZE = 256;
+ char buffer[SIZE];
+ String8 result;
+
+ snprintf(buffer, SIZE, "\nAudioPolicyManager Dump: %p\n", this);
+ result.append(buffer);
+
+ snprintf(buffer, SIZE, " Primary Output: %d\n", mPrimaryOutput);
+ result.append(buffer);
+ snprintf(buffer, SIZE, " Phone state: %d\n", mPhoneState);
+ result.append(buffer);
+ snprintf(buffer, SIZE, " Force use for communications %d\n",
+ mForceUse[AUDIO_POLICY_FORCE_FOR_COMMUNICATION]);
+ result.append(buffer);
+ snprintf(buffer, SIZE, " Force use for media %d\n", mForceUse[AUDIO_POLICY_FORCE_FOR_MEDIA]);
+ result.append(buffer);
+ snprintf(buffer, SIZE, " Force use for record %d\n", mForceUse[AUDIO_POLICY_FORCE_FOR_RECORD]);
+ result.append(buffer);
+ snprintf(buffer, SIZE, " Force use for dock %d\n", mForceUse[AUDIO_POLICY_FORCE_FOR_DOCK]);
+ result.append(buffer);
+ snprintf(buffer, SIZE, " Force use for system %d\n", mForceUse[AUDIO_POLICY_FORCE_FOR_SYSTEM]);
+ result.append(buffer);
+ snprintf(buffer, SIZE, " Force use for hdmi system audio %d\n",
+ mForceUse[AUDIO_POLICY_FORCE_FOR_HDMI_SYSTEM_AUDIO]);
+ result.append(buffer);
+
+ snprintf(buffer, SIZE, " Available output devices:\n");
+ result.append(buffer);
+ write(fd, result.string(), result.size());
+ for (size_t i = 0; i < mAvailableOutputDevices.size(); i++) {
+ mAvailableOutputDevices[i]->dump(fd, 2, i);
+ }
+ snprintf(buffer, SIZE, "\n Available input devices:\n");
+ write(fd, buffer, strlen(buffer));
+ for (size_t i = 0; i < mAvailableInputDevices.size(); i++) {
+ mAvailableInputDevices[i]->dump(fd, 2, i);
+ }
+
+ snprintf(buffer, SIZE, "\nHW Modules dump:\n");
+ write(fd, buffer, strlen(buffer));
+ for (size_t i = 0; i < mHwModules.size(); i++) {
+ snprintf(buffer, SIZE, "- HW Module %zu:\n", i + 1);
+ write(fd, buffer, strlen(buffer));
+ mHwModules[i]->dump(fd);
+ }
+
+ snprintf(buffer, SIZE, "\nOutputs dump:\n");
+ write(fd, buffer, strlen(buffer));
+ for (size_t i = 0; i < mOutputs.size(); i++) {
+ snprintf(buffer, SIZE, "- Output %d dump:\n", mOutputs.keyAt(i));
+ write(fd, buffer, strlen(buffer));
+ mOutputs.valueAt(i)->dump(fd);
+ }
+
+ snprintf(buffer, SIZE, "\nInputs dump:\n");
+ write(fd, buffer, strlen(buffer));
+ for (size_t i = 0; i < mInputs.size(); i++) {
+ snprintf(buffer, SIZE, "- Input %d dump:\n", mInputs.keyAt(i));
+ write(fd, buffer, strlen(buffer));
+ mInputs.valueAt(i)->dump(fd);
+ }
+
+ snprintf(buffer, SIZE, "\nStreams dump:\n");
+ write(fd, buffer, strlen(buffer));
+ snprintf(buffer, SIZE,
+ " Stream Can be muted Index Min Index Max Index Cur [device : index]...\n");
+ write(fd, buffer, strlen(buffer));
+ for (size_t i = 0; i < AUDIO_STREAM_CNT; i++) {
+ snprintf(buffer, SIZE, " %02zu ", i);
+ write(fd, buffer, strlen(buffer));
+ mStreams[i].dump(fd);
+ }
+
+ snprintf(buffer, SIZE, "\nTotal Effects CPU: %f MIPS, Total Effects memory: %d KB\n",
+ (float)mTotalEffectsCpuLoad/10, mTotalEffectsMemory);
+ write(fd, buffer, strlen(buffer));
+
+ snprintf(buffer, SIZE, "Registered effects:\n");
+ write(fd, buffer, strlen(buffer));
+ for (size_t i = 0; i < mEffects.size(); i++) {
+ snprintf(buffer, SIZE, "- Effect %d dump:\n", mEffects.keyAt(i));
+ write(fd, buffer, strlen(buffer));
+ mEffects.valueAt(i)->dump(fd);
+ }
+
+ snprintf(buffer, SIZE, "\nAudio Patches:\n");
+ write(fd, buffer, strlen(buffer));
+ for (size_t i = 0; i < mAudioPatches.size(); i++) {
+ mAudioPatches[i]->dump(fd, 2, i);
+ }
+
+ return NO_ERROR;
+}
+
+// This function checks for the parameters which can be offloaded.
+// This can be enhanced depending on the capability of the DSP and policy
+// of the system.
+bool AudioPolicyManager::isOffloadSupported(const audio_offload_info_t& offloadInfo)
+{
+ ALOGV("isOffloadSupported: SR=%u, CM=0x%x, Format=0x%x, StreamType=%d,"
+ " BitRate=%u, duration=%" PRId64 " us, has_video=%d",
+ offloadInfo.sample_rate, offloadInfo.channel_mask,
+ offloadInfo.format,
+ offloadInfo.stream_type, offloadInfo.bit_rate, offloadInfo.duration_us,
+ offloadInfo.has_video);
+
+ // Check if offload has been disabled
+ char propValue[PROPERTY_VALUE_MAX];
+ if (property_get("audio.offload.disable", propValue, "0")) {
+ if (atoi(propValue) != 0) {
+ ALOGV("offload disabled by audio.offload.disable=%s", propValue );
+ return false;
+ }
+ }
+
+ // Check if stream type is music, then only allow offload as of now.
+ if (offloadInfo.stream_type != AUDIO_STREAM_MUSIC)
+ {
+ ALOGV("isOffloadSupported: stream_type != MUSIC, returning false");
+ return false;
+ }
+
+ //TODO: enable audio offloading with video when ready
+ if (offloadInfo.has_video)
+ {
+ ALOGV("isOffloadSupported: has_video == true, returning false");
+ return false;
+ }
+
+ //If duration is less than minimum value defined in property, return false
+ if (property_get("audio.offload.min.duration.secs", propValue, NULL)) {
+ if (offloadInfo.duration_us < (atoi(propValue) * 1000000 )) {
+ ALOGV("Offload denied by duration < audio.offload.min.duration.secs(=%s)", propValue);
+ return false;
+ }
+ } else if (offloadInfo.duration_us < OFFLOAD_DEFAULT_MIN_DURATION_SECS * 1000000) {
+ ALOGV("Offload denied by duration < default min(=%u)", OFFLOAD_DEFAULT_MIN_DURATION_SECS);
+ return false;
+ }
+
+ // Do not allow offloading if one non offloadable effect is enabled. This prevents from
+ // creating an offloaded track and tearing it down immediately after start when audioflinger
+ // detects there is an active non offloadable effect.
+ // FIXME: We should check the audio session here but we do not have it in this context.
+ // This may prevent offloading in rare situations where effects are left active by apps
+ // in the background.
+ if (isNonOffloadableEffectEnabled()) {
+ return false;
+ }
+
+ // See if there is a profile to support this.
+ // AUDIO_DEVICE_NONE
+ sp<IOProfile> profile = getProfileForDirectOutput(AUDIO_DEVICE_NONE /*ignore device */,
+ offloadInfo.sample_rate,
+ offloadInfo.format,
+ offloadInfo.channel_mask,
+ AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD);
+ ALOGV("isOffloadSupported() profile %sfound", profile != 0 ? "" : "NOT ");
+ return (profile != 0);
+}
+
+status_t AudioPolicyManager::listAudioPorts(audio_port_role_t role,
+ audio_port_type_t type,
+ unsigned int *num_ports,
+ struct audio_port *ports,
+ unsigned int *generation)
+{
+ if (num_ports == NULL || (*num_ports != 0 && ports == NULL) ||
+ generation == NULL) {
+ return BAD_VALUE;
+ }
+ ALOGV("listAudioPorts() role %d type %d num_ports %d ports %p", role, type, *num_ports, ports);
+ if (ports == NULL) {
+ *num_ports = 0;
+ }
+
+ size_t portsWritten = 0;
+ size_t portsMax = *num_ports;
+ *num_ports = 0;
+ if (type == AUDIO_PORT_TYPE_NONE || type == AUDIO_PORT_TYPE_DEVICE) {
+ if (role == AUDIO_PORT_ROLE_SINK || role == AUDIO_PORT_ROLE_NONE) {
+ for (size_t i = 0;
+ i < mAvailableOutputDevices.size() && portsWritten < portsMax; i++) {
+ mAvailableOutputDevices[i]->toAudioPort(&ports[portsWritten++]);
+ }
+ *num_ports += mAvailableOutputDevices.size();
+ }
+ if (role == AUDIO_PORT_ROLE_SOURCE || role == AUDIO_PORT_ROLE_NONE) {
+ for (size_t i = 0;
+ i < mAvailableInputDevices.size() && portsWritten < portsMax; i++) {
+ mAvailableInputDevices[i]->toAudioPort(&ports[portsWritten++]);
+ }
+ *num_ports += mAvailableInputDevices.size();
+ }
+ }
+ if (type == AUDIO_PORT_TYPE_NONE || type == AUDIO_PORT_TYPE_MIX) {
+ if (role == AUDIO_PORT_ROLE_SINK || role == AUDIO_PORT_ROLE_NONE) {
+ for (size_t i = 0; i < mInputs.size() && portsWritten < portsMax; i++) {
+ mInputs[i]->toAudioPort(&ports[portsWritten++]);
+ }
+ *num_ports += mInputs.size();
+ }
+ if (role == AUDIO_PORT_ROLE_SOURCE || role == AUDIO_PORT_ROLE_NONE) {
+ size_t numOutputs = 0;
+ for (size_t i = 0; i < mOutputs.size(); i++) {
+ if (!mOutputs[i]->isDuplicated()) {
+ numOutputs++;
+ if (portsWritten < portsMax) {
+ mOutputs[i]->toAudioPort(&ports[portsWritten++]);
+ }
+ }
+ }
+ *num_ports += numOutputs;
+ }
+ }
+ *generation = curAudioPortGeneration();
+ ALOGV("listAudioPorts() got %zu ports needed %d", portsWritten, *num_ports);
+ return NO_ERROR;
+}
+
+status_t AudioPolicyManager::getAudioPort(struct audio_port *port __unused)
+{
+ return NO_ERROR;
+}
+
+sp<AudioOutputDescriptor> AudioPolicyManager::getOutputFromId(
+ audio_port_handle_t id) const
+{
+ sp<AudioOutputDescriptor> outputDesc = NULL;
+ for (size_t i = 0; i < mOutputs.size(); i++) {
+ outputDesc = mOutputs.valueAt(i);
+ if (outputDesc->mId == id) {
+ break;
+ }
+ }
+ return outputDesc;
+}
+
+sp<AudioInputDescriptor> AudioPolicyManager::getInputFromId(
+ audio_port_handle_t id) const
+{
+ sp<AudioInputDescriptor> inputDesc = NULL;
+ for (size_t i = 0; i < mInputs.size(); i++) {
+ inputDesc = mInputs.valueAt(i);
+ if (inputDesc->mId == id) {
+ break;
+ }
+ }
+ return inputDesc;
+}
+
+sp <HwModule> AudioPolicyManager::getModuleForDevice(
+ audio_devices_t device) const
+{
+ sp <HwModule> module;
+
+ for (size_t i = 0; i < mHwModules.size(); i++) {
+ if (mHwModules[i]->mHandle == 0) {
+ continue;
+ }
+ if (audio_is_output_device(device)) {
+ for (size_t j = 0; j < mHwModules[i]->mOutputProfiles.size(); j++)
+ {
+ if (mHwModules[i]->mOutputProfiles[j]->mSupportedDevices.types() & device) {
+ return mHwModules[i];
+ }
+ }
+ } else {
+ for (size_t j = 0; j < mHwModules[i]->mInputProfiles.size(); j++) {
+ if (mHwModules[i]->mInputProfiles[j]->mSupportedDevices.types() &
+ device & ~AUDIO_DEVICE_BIT_IN) {
+ return mHwModules[i];
+ }
+ }
+ }
+ }
+ return module;
+}
+
+sp <HwModule> AudioPolicyManager::getModuleFromName(const char *name) const
+{
+ sp <HwModule> module;
+
+ for (size_t i = 0; i < mHwModules.size(); i++)
+ {
+ if (strcmp(mHwModules[i]->mName, name) == 0) {
+ return mHwModules[i];
+ }
+ }
+ return module;
+}
+
+audio_devices_t AudioPolicyManager::availablePrimaryOutputDevices()
+{
+ sp<AudioOutputDescriptor> outputDesc = mOutputs.valueFor(mPrimaryOutput);
+ audio_devices_t devices = outputDesc->mProfile->mSupportedDevices.types();
+ return devices & mAvailableOutputDevices.types();
+}
+
+audio_devices_t AudioPolicyManager::availablePrimaryInputDevices()
+{
+ audio_module_handle_t primaryHandle =
+ mOutputs.valueFor(mPrimaryOutput)->mProfile->mModule->mHandle;
+ audio_devices_t devices = AUDIO_DEVICE_NONE;
+ for (size_t i = 0; i < mAvailableInputDevices.size(); i++) {
+ if (mAvailableInputDevices[i]->mModule->mHandle == primaryHandle) {
+ devices |= mAvailableInputDevices[i]->mDeviceType;
+ }
+ }
+ return devices;
+}
+
+status_t AudioPolicyManager::createAudioPatch(const struct audio_patch *patch,
+ audio_patch_handle_t *handle,
+ uid_t uid)
+{
+ ALOGV("createAudioPatch()");
+
+ if (handle == NULL || patch == NULL) {
+ return BAD_VALUE;
+ }
+ ALOGV("createAudioPatch() num sources %d num sinks %d", patch->num_sources, patch->num_sinks);
+
+ if (patch->num_sources == 0 || patch->num_sources > AUDIO_PATCH_PORTS_MAX ||
+ patch->num_sinks == 0 || patch->num_sinks > AUDIO_PATCH_PORTS_MAX) {
+ return BAD_VALUE;
+ }
+ // only one source per audio patch supported for now
+ if (patch->num_sources > 1) {
+ return INVALID_OPERATION;
+ }
+
+ if (patch->sources[0].role != AUDIO_PORT_ROLE_SOURCE) {
+ return INVALID_OPERATION;
+ }
+ for (size_t i = 0; i < patch->num_sinks; i++) {
+ if (patch->sinks[i].role != AUDIO_PORT_ROLE_SINK) {
+ return INVALID_OPERATION;
+ }
+ }
+
+ sp<AudioPatch> patchDesc;
+ ssize_t index = mAudioPatches.indexOfKey(*handle);
+
+ ALOGV("createAudioPatch source id %d role %d type %d", patch->sources[0].id,
+ patch->sources[0].role,
+ patch->sources[0].type);
+#if LOG_NDEBUG == 0
+ for (size_t i = 0; i < patch->num_sinks; i++) {
+ ALOGV("createAudioPatch sink %d: id %d role %d type %d", i, patch->sinks[i].id,
+ patch->sinks[i].role,
+ patch->sinks[i].type);
+ }
+#endif
+
+ if (index >= 0) {
+ patchDesc = mAudioPatches.valueAt(index);
+ ALOGV("createAudioPatch() mUidCached %d patchDesc->mUid %d uid %d",
+ mUidCached, patchDesc->mUid, uid);
+ if (patchDesc->mUid != mUidCached && uid != patchDesc->mUid) {
+ return INVALID_OPERATION;
+ }
+ } else {
+ *handle = 0;
+ }
+
+ if (patch->sources[0].type == AUDIO_PORT_TYPE_MIX) {
+ sp<AudioOutputDescriptor> outputDesc = getOutputFromId(patch->sources[0].id);
+ if (outputDesc == NULL) {
+ ALOGV("createAudioPatch() output not found for id %d", patch->sources[0].id);
+ return BAD_VALUE;
+ }
+ ALOG_ASSERT(!outputDesc->isDuplicated(),"duplicated output %d in source in ports",
+ outputDesc->mIoHandle);
+ if (patchDesc != 0) {
+ if (patchDesc->mPatch.sources[0].id != patch->sources[0].id) {
+ ALOGV("createAudioPatch() source id differs for patch current id %d new id %d",
+ patchDesc->mPatch.sources[0].id, patch->sources[0].id);
+ return BAD_VALUE;
+ }
+ }
+ DeviceVector devices;
+ for (size_t i = 0; i < patch->num_sinks; i++) {
+ // Only support mix to devices connection
+ // TODO add support for mix to mix connection
+ if (patch->sinks[i].type != AUDIO_PORT_TYPE_DEVICE) {
+ ALOGV("createAudioPatch() source mix but sink is not a device");
+ return INVALID_OPERATION;
+ }
+ sp<DeviceDescriptor> devDesc =
+ mAvailableOutputDevices.getDeviceFromId(patch->sinks[i].id);
+ if (devDesc == 0) {
+ ALOGV("createAudioPatch() out device not found for id %d", patch->sinks[i].id);
+ return BAD_VALUE;
+ }
+
+ if (!outputDesc->mProfile->isCompatibleProfile(devDesc->mDeviceType,
+ devDesc->mAddress,
+ patch->sources[0].sample_rate,
+ NULL, // updatedSamplingRate
+ patch->sources[0].format,
+ patch->sources[0].channel_mask,
+ AUDIO_OUTPUT_FLAG_NONE /*FIXME*/)) {
+ ALOGV("createAudioPatch() profile not supported for device %08x",
+ devDesc->mDeviceType);
+ return INVALID_OPERATION;
+ }
+ devices.add(devDesc);
+ }
+ if (devices.size() == 0) {
+ return INVALID_OPERATION;
+ }
+
+ // TODO: reconfigure output format and channels here
+ ALOGV("createAudioPatch() setting device %08x on output %d",
+ devices.types(), outputDesc->mIoHandle);
+ setOutputDevice(outputDesc->mIoHandle, devices.types(), true, 0, handle);
+ index = mAudioPatches.indexOfKey(*handle);
+ if (index >= 0) {
+ if (patchDesc != 0 && patchDesc != mAudioPatches.valueAt(index)) {
+ ALOGW("createAudioPatch() setOutputDevice() did not reuse the patch provided");
+ }
+ patchDesc = mAudioPatches.valueAt(index);
+ patchDesc->mUid = uid;
+ ALOGV("createAudioPatch() success");
+ } else {
+ ALOGW("createAudioPatch() setOutputDevice() failed to create a patch");
+ return INVALID_OPERATION;
+ }
+ } else if (patch->sources[0].type == AUDIO_PORT_TYPE_DEVICE) {
+ if (patch->sinks[0].type == AUDIO_PORT_TYPE_MIX) {
+ // input device to input mix connection
+ // only one sink supported when connecting an input device to a mix
+ if (patch->num_sinks > 1) {
+ return INVALID_OPERATION;
+ }
+ sp<AudioInputDescriptor> inputDesc = getInputFromId(patch->sinks[0].id);
+ if (inputDesc == NULL) {
+ return BAD_VALUE;
+ }
+ if (patchDesc != 0) {
+ if (patchDesc->mPatch.sinks[0].id != patch->sinks[0].id) {
+ return BAD_VALUE;
+ }
+ }
+ sp<DeviceDescriptor> devDesc =
+ mAvailableInputDevices.getDeviceFromId(patch->sources[0].id);
+ if (devDesc == 0) {
+ return BAD_VALUE;
+ }
+
+ if (!inputDesc->mProfile->isCompatibleProfile(devDesc->mDeviceType,
+ devDesc->mAddress,
+ patch->sinks[0].sample_rate,
+ NULL, /*updatedSampleRate*/
+ patch->sinks[0].format,
+ patch->sinks[0].channel_mask,
+ // FIXME for the parameter type,
+ // and the NONE
+ (audio_output_flags_t)
+ AUDIO_INPUT_FLAG_NONE)) {
+ return INVALID_OPERATION;
+ }
+ // TODO: reconfigure output format and channels here
+ ALOGV("createAudioPatch() setting device %08x on output %d",
+ devDesc->mDeviceType, inputDesc->mIoHandle);
+ setInputDevice(inputDesc->mIoHandle, devDesc->mDeviceType, true, handle);
+ index = mAudioPatches.indexOfKey(*handle);
+ if (index >= 0) {
+ if (patchDesc != 0 && patchDesc != mAudioPatches.valueAt(index)) {
+ ALOGW("createAudioPatch() setInputDevice() did not reuse the patch provided");
+ }
+ patchDesc = mAudioPatches.valueAt(index);
+ patchDesc->mUid = uid;
+ ALOGV("createAudioPatch() success");
+ } else {
+ ALOGW("createAudioPatch() setInputDevice() failed to create a patch");
+ return INVALID_OPERATION;
+ }
+ } else if (patch->sinks[0].type == AUDIO_PORT_TYPE_DEVICE) {
+ // device to device connection
+ if (patchDesc != 0) {
+ if (patchDesc->mPatch.sources[0].id != patch->sources[0].id) {
+ return BAD_VALUE;
+ }
+ }
+ sp<DeviceDescriptor> srcDeviceDesc =
+ mAvailableInputDevices.getDeviceFromId(patch->sources[0].id);
+ if (srcDeviceDesc == 0) {
+ return BAD_VALUE;
+ }
+
+ //update source and sink with our own data as the data passed in the patch may
+ // be incomplete.
+ struct audio_patch newPatch = *patch;
+ srcDeviceDesc->toAudioPortConfig(&newPatch.sources[0], &patch->sources[0]);
+
+ for (size_t i = 0; i < patch->num_sinks; i++) {
+ if (patch->sinks[i].type != AUDIO_PORT_TYPE_DEVICE) {
+ ALOGV("createAudioPatch() source device but one sink is not a device");
+ return INVALID_OPERATION;
+ }
+
+ sp<DeviceDescriptor> sinkDeviceDesc =
+ mAvailableOutputDevices.getDeviceFromId(patch->sinks[i].id);
+ if (sinkDeviceDesc == 0) {
+ return BAD_VALUE;
+ }
+ sinkDeviceDesc->toAudioPortConfig(&newPatch.sinks[i], &patch->sinks[i]);
+
+ if (srcDeviceDesc->mModule != sinkDeviceDesc->mModule) {
+ // only one sink supported when connected devices across HW modules
+ if (patch->num_sinks > 1) {
+ return INVALID_OPERATION;
+ }
+ SortedVector<audio_io_handle_t> outputs =
+ getOutputsForDevice(sinkDeviceDesc->mDeviceType,
+ mOutputs);
+ // if the sink device is reachable via an opened output stream, request to go via
+ // this output stream by adding a second source to the patch description
+ audio_io_handle_t output = selectOutput(outputs,
+ AUDIO_OUTPUT_FLAG_NONE,
+ AUDIO_FORMAT_INVALID);
+ if (output != AUDIO_IO_HANDLE_NONE) {
+ sp<AudioOutputDescriptor> outputDesc = mOutputs.valueFor(output);
+ if (outputDesc->isDuplicated()) {
+ return INVALID_OPERATION;
+ }
+ outputDesc->toAudioPortConfig(&newPatch.sources[1], &patch->sources[0]);
+ newPatch.num_sources = 2;
+ }
+ }
+ }
+ // TODO: check from routing capabilities in config file and other conflicting patches
+
+ audio_patch_handle_t afPatchHandle = AUDIO_PATCH_HANDLE_NONE;
+ if (index >= 0) {
+ afPatchHandle = patchDesc->mAfPatchHandle;
+ }
+
+ status_t status = mpClientInterface->createAudioPatch(&newPatch,
+ &afPatchHandle,
+ 0);
+ ALOGV("createAudioPatch() patch panel returned %d patchHandle %d",
+ status, afPatchHandle);
+ if (status == NO_ERROR) {
+ if (index < 0) {
+ patchDesc = new AudioPatch((audio_patch_handle_t)nextUniqueId(),
+ &newPatch, uid);
+ addAudioPatch(patchDesc->mHandle, patchDesc);
+ } else {
+ patchDesc->mPatch = newPatch;
+ }
+ patchDesc->mAfPatchHandle = afPatchHandle;
+ *handle = patchDesc->mHandle;
+ nextAudioPortGeneration();
+ mpClientInterface->onAudioPatchListUpdate();
+ } else {
+ ALOGW("createAudioPatch() patch panel could not connect device patch, error %d",
+ status);
+ return INVALID_OPERATION;
+ }
+ } else {
+ return BAD_VALUE;
+ }
+ } else {
+ return BAD_VALUE;
+ }
+ return NO_ERROR;
+}
+
+status_t AudioPolicyManager::releaseAudioPatch(audio_patch_handle_t handle,
+ uid_t uid)
+{
+ ALOGV("releaseAudioPatch() patch %d", handle);
+
+ ssize_t index = mAudioPatches.indexOfKey(handle);
+
+ if (index < 0) {
+ return BAD_VALUE;
+ }
+ sp<AudioPatch> patchDesc = mAudioPatches.valueAt(index);
+ ALOGV("releaseAudioPatch() mUidCached %d patchDesc->mUid %d uid %d",
+ mUidCached, patchDesc->mUid, uid);
+ if (patchDesc->mUid != mUidCached && uid != patchDesc->mUid) {
+ return INVALID_OPERATION;
+ }
+
+ struct audio_patch *patch = &patchDesc->mPatch;
+ patchDesc->mUid = mUidCached;
+ if (patch->sources[0].type == AUDIO_PORT_TYPE_MIX) {
+ sp<AudioOutputDescriptor> outputDesc = getOutputFromId(patch->sources[0].id);
+ if (outputDesc == NULL) {
+ ALOGV("releaseAudioPatch() output not found for id %d", patch->sources[0].id);
+ return BAD_VALUE;
+ }
+
+ setOutputDevice(outputDesc->mIoHandle,
+ getNewOutputDevice(outputDesc->mIoHandle, true /*fromCache*/),
+ true,
+ 0,
+ NULL);
+ } else if (patch->sources[0].type == AUDIO_PORT_TYPE_DEVICE) {
+ if (patch->sinks[0].type == AUDIO_PORT_TYPE_MIX) {
+ sp<AudioInputDescriptor> inputDesc = getInputFromId(patch->sinks[0].id);
+ if (inputDesc == NULL) {
+ ALOGV("releaseAudioPatch() input not found for id %d", patch->sinks[0].id);
+ return BAD_VALUE;
+ }
+ setInputDevice(inputDesc->mIoHandle,
+ getNewInputDevice(inputDesc->mIoHandle),
+ true,
+ NULL);
+ } else if (patch->sinks[0].type == AUDIO_PORT_TYPE_DEVICE) {
+ audio_patch_handle_t afPatchHandle = patchDesc->mAfPatchHandle;
+ status_t status = mpClientInterface->releaseAudioPatch(patchDesc->mAfPatchHandle, 0);
+ ALOGV("releaseAudioPatch() patch panel returned %d patchHandle %d",
+ status, patchDesc->mAfPatchHandle);
+ removeAudioPatch(patchDesc->mHandle);
+ nextAudioPortGeneration();
+ mpClientInterface->onAudioPatchListUpdate();
+ } else {
+ return BAD_VALUE;
+ }
+ } else {
+ return BAD_VALUE;
+ }
+ return NO_ERROR;
+}
+
+status_t AudioPolicyManager::listAudioPatches(unsigned int *num_patches,
+ struct audio_patch *patches,
+ unsigned int *generation)
+{
+ if (num_patches == NULL || (*num_patches != 0 && patches == NULL) ||
+ generation == NULL) {
+ return BAD_VALUE;
+ }
+ ALOGV("listAudioPatches() num_patches %d patches %p available patches %zu",
+ *num_patches, patches, mAudioPatches.size());
+ if (patches == NULL) {
+ *num_patches = 0;
+ }
+
+ size_t patchesWritten = 0;
+ size_t patchesMax = *num_patches;
+ for (size_t i = 0;
+ i < mAudioPatches.size() && patchesWritten < patchesMax; i++) {
+ patches[patchesWritten] = mAudioPatches[i]->mPatch;
+ patches[patchesWritten++].id = mAudioPatches[i]->mHandle;
+ ALOGV("listAudioPatches() patch %zu num_sources %d num_sinks %d",
+ i, mAudioPatches[i]->mPatch.num_sources, mAudioPatches[i]->mPatch.num_sinks);
+ }
+ *num_patches = mAudioPatches.size();
+
+ *generation = curAudioPortGeneration();
+ ALOGV("listAudioPatches() got %zu patches needed %d", patchesWritten, *num_patches);
+ return NO_ERROR;
+}
+
+status_t AudioPolicyManager::setAudioPortConfig(const struct audio_port_config *config)
+{
+ ALOGV("setAudioPortConfig()");
+
+ if (config == NULL) {
+ return BAD_VALUE;
+ }
+ ALOGV("setAudioPortConfig() on port handle %d", config->id);
+ // Only support gain configuration for now
+ if (config->config_mask != AUDIO_PORT_CONFIG_GAIN) {
+ return INVALID_OPERATION;
+ }
+
+ sp<AudioPortConfig> audioPortConfig;
+ if (config->type == AUDIO_PORT_TYPE_MIX) {
+ if (config->role == AUDIO_PORT_ROLE_SOURCE) {
+ sp<AudioOutputDescriptor> outputDesc = getOutputFromId(config->id);
+ if (outputDesc == NULL) {
+ return BAD_VALUE;
+ }
+ ALOG_ASSERT(!outputDesc->isDuplicated(),
+ "setAudioPortConfig() called on duplicated output %d",
+ outputDesc->mIoHandle);
+ audioPortConfig = outputDesc;
+ } else if (config->role == AUDIO_PORT_ROLE_SINK) {
+ sp<AudioInputDescriptor> inputDesc = getInputFromId(config->id);
+ if (inputDesc == NULL) {
+ return BAD_VALUE;
+ }
+ audioPortConfig = inputDesc;
+ } else {
+ return BAD_VALUE;
+ }
+ } else if (config->type == AUDIO_PORT_TYPE_DEVICE) {
+ sp<DeviceDescriptor> deviceDesc;
+ if (config->role == AUDIO_PORT_ROLE_SOURCE) {
+ deviceDesc = mAvailableInputDevices.getDeviceFromId(config->id);
+ } else if (config->role == AUDIO_PORT_ROLE_SINK) {
+ deviceDesc = mAvailableOutputDevices.getDeviceFromId(config->id);
+ } else {
+ return BAD_VALUE;
+ }
+ if (deviceDesc == NULL) {
+ return BAD_VALUE;
+ }
+ audioPortConfig = deviceDesc;
+ } else {
+ return BAD_VALUE;
+ }
+
+ struct audio_port_config backupConfig;
+ status_t status = audioPortConfig->applyAudioPortConfig(config, &backupConfig);
+ if (status == NO_ERROR) {
+ struct audio_port_config newConfig;
+ audioPortConfig->toAudioPortConfig(&newConfig, config);
+ status = mpClientInterface->setAudioPortConfig(&newConfig, 0);
+ }
+ if (status != NO_ERROR) {
+ audioPortConfig->applyAudioPortConfig(&backupConfig);
+ }
+
+ return status;
+}
+
+void AudioPolicyManager::clearAudioPatches(uid_t uid)
+{
+ for (ssize_t i = (ssize_t)mAudioPatches.size() - 1; i >= 0; i--) {
+ sp<AudioPatch> patchDesc = mAudioPatches.valueAt(i);
+ if (patchDesc->mUid == uid) {
+ releaseAudioPatch(mAudioPatches.keyAt(i), uid);
+ }
+ }
+}
+
+status_t AudioPolicyManager::acquireSoundTriggerSession(audio_session_t *session,
+ audio_io_handle_t *ioHandle,
+ audio_devices_t *device)
+{
+ *session = (audio_session_t)mpClientInterface->newAudioUniqueId();
+ *ioHandle = (audio_io_handle_t)mpClientInterface->newAudioUniqueId();
+ *device = getDeviceAndMixForInputSource(AUDIO_SOURCE_HOTWORD);
+
+ mSoundTriggerSessions.add(*session, *ioHandle);
+
+ return NO_ERROR;
+}
+
+status_t AudioPolicyManager::releaseSoundTriggerSession(audio_session_t session)
+{
+ ssize_t index = mSoundTriggerSessions.indexOfKey(session);
+ if (index < 0) {
+ ALOGW("acquireSoundTriggerSession() session %d not registered", session);
+ return BAD_VALUE;
+ }
+
+ mSoundTriggerSessions.removeItem(session);
+ return NO_ERROR;
+}
+
+status_t AudioPolicyManager::addAudioPatch(audio_patch_handle_t handle,
+ const sp<AudioPatch>& patch)
+{
+ ssize_t index = mAudioPatches.indexOfKey(handle);
+
+ if (index >= 0) {
+ ALOGW("addAudioPatch() patch %d already in", handle);
+ return ALREADY_EXISTS;
+ }
+ mAudioPatches.add(handle, patch);
+ ALOGV("addAudioPatch() handle %d af handle %d num_sources %d num_sinks %d source handle %d"
+ "sink handle %d",
+ handle, patch->mAfPatchHandle, patch->mPatch.num_sources, patch->mPatch.num_sinks,
+ patch->mPatch.sources[0].id, patch->mPatch.sinks[0].id);
+ return NO_ERROR;
+}
+
+status_t AudioPolicyManager::removeAudioPatch(audio_patch_handle_t handle)
+{
+ ssize_t index = mAudioPatches.indexOfKey(handle);
+
+ if (index < 0) {
+ ALOGW("removeAudioPatch() patch %d not in", handle);
+ return ALREADY_EXISTS;
+ }
+ ALOGV("removeAudioPatch() handle %d af handle %d", handle,
+ mAudioPatches.valueAt(index)->mAfPatchHandle);
+ mAudioPatches.removeItemsAt(index);
+ return NO_ERROR;
+}
+
+// ----------------------------------------------------------------------------
+// AudioPolicyManager
+// ----------------------------------------------------------------------------
+
+uint32_t AudioPolicyManager::nextUniqueId()
+{
+ return android_atomic_inc(&mNextUniqueId);
+}
+
+uint32_t AudioPolicyManager::nextAudioPortGeneration()
+{
+ return android_atomic_inc(&mAudioPortGeneration);
+}
+
+int32_t volatile AudioPolicyManager::mNextUniqueId = 1;
+
+AudioPolicyManager::AudioPolicyManager(AudioPolicyClientInterface *clientInterface)
+ :
+#ifdef AUDIO_POLICY_TEST
+ Thread(false),
+#endif //AUDIO_POLICY_TEST
+ mPrimaryOutput((audio_io_handle_t)0),
+ mPhoneState(AUDIO_MODE_NORMAL),
+ mLimitRingtoneVolume(false), mLastVoiceVolume(-1.0f),
+ mTotalEffectsCpuLoad(0), mTotalEffectsMemory(0),
+ mA2dpSuspended(false),
+ mSpeakerDrcEnabled(false),
+ mAudioPortGeneration(1),
+ mBeaconMuteRefCount(0),
+ mBeaconPlayingRefCount(0),
+ mBeaconMuted(false)
+{
+ mUidCached = getuid();
+ mpClientInterface = clientInterface;
+
+ for (int i = 0; i < AUDIO_POLICY_FORCE_USE_CNT; i++) {
+ mForceUse[i] = AUDIO_POLICY_FORCE_NONE;
+ }
+
+ mDefaultOutputDevice = new DeviceDescriptor(String8("Speaker"), AUDIO_DEVICE_OUT_SPEAKER);
+ if (loadAudioPolicyConfig(AUDIO_POLICY_VENDOR_CONFIG_FILE) != NO_ERROR) {
+ if (loadAudioPolicyConfig(AUDIO_POLICY_CONFIG_FILE) != NO_ERROR) {
+ ALOGE("could not load audio policy configuration file, setting defaults");
+ defaultAudioPolicyConfig();
+ }
+ }
+ // mAvailableOutputDevices and mAvailableInputDevices now contain all attached devices
+
+ // must be done after reading the policy
+ initializeVolumeCurves();
+
+ // open all output streams needed to access attached devices
+ audio_devices_t outputDeviceTypes = mAvailableOutputDevices.types();
+ audio_devices_t inputDeviceTypes = mAvailableInputDevices.types() & ~AUDIO_DEVICE_BIT_IN;
+ for (size_t i = 0; i < mHwModules.size(); i++) {
+ mHwModules[i]->mHandle = mpClientInterface->loadHwModule(mHwModules[i]->mName);
+ if (mHwModules[i]->mHandle == 0) {
+ ALOGW("could not open HW module %s", mHwModules[i]->mName);
+ continue;
+ }
+ // open all output streams needed to access attached devices
+ // except for direct output streams that are only opened when they are actually
+ // required by an app.
+ // This also validates mAvailableOutputDevices list
+ for (size_t j = 0; j < mHwModules[i]->mOutputProfiles.size(); j++)
+ {
+ const sp<IOProfile> outProfile = mHwModules[i]->mOutputProfiles[j];
+
+ if (outProfile->mSupportedDevices.isEmpty()) {
+ ALOGW("Output profile contains no device on module %s", mHwModules[i]->mName);
+ continue;
+ }
+
+ if ((outProfile->mFlags & AUDIO_OUTPUT_FLAG_DIRECT) != 0) {
+ continue;
+ }
+ audio_devices_t profileType = outProfile->mSupportedDevices.types();
+ if ((profileType & mDefaultOutputDevice->mDeviceType) != AUDIO_DEVICE_NONE) {
+ profileType = mDefaultOutputDevice->mDeviceType;
+ } else {
+ // chose first device present in mSupportedDevices also part of
+ // outputDeviceTypes
+ for (size_t k = 0; k < outProfile->mSupportedDevices.size(); k++) {
+ profileType = outProfile->mSupportedDevices[k]->mDeviceType;
+ if ((profileType & outputDeviceTypes) != 0) {
+ break;
+ }
+ }
+ }
+ if ((profileType & outputDeviceTypes) == 0) {
+ continue;
+ }
+ sp<AudioOutputDescriptor> outputDesc = new AudioOutputDescriptor(outProfile);
+
+ outputDesc->mDevice = profileType;
+ audio_config_t config = AUDIO_CONFIG_INITIALIZER;
+ config.sample_rate = outputDesc->mSamplingRate;
+ config.channel_mask = outputDesc->mChannelMask;
+ config.format = outputDesc->mFormat;
+ audio_io_handle_t output = AUDIO_IO_HANDLE_NONE;
+ status_t status = mpClientInterface->openOutput(outProfile->mModule->mHandle,
+ &output,
+ &config,
+ &outputDesc->mDevice,
+ String8(""),
+ &outputDesc->mLatency,
+ outputDesc->mFlags);
+
+ if (status != NO_ERROR) {
+ ALOGW("Cannot open output stream for device %08x on hw module %s",
+ outputDesc->mDevice,
+ mHwModules[i]->mName);
+ } else {
+ outputDesc->mSamplingRate = config.sample_rate;
+ outputDesc->mChannelMask = config.channel_mask;
+ outputDesc->mFormat = config.format;
+
+ for (size_t k = 0; k < outProfile->mSupportedDevices.size(); k++) {
+ audio_devices_t type = outProfile->mSupportedDevices[k]->mDeviceType;
+ ssize_t index =
+ mAvailableOutputDevices.indexOf(outProfile->mSupportedDevices[k]);
+ // give a valid ID to an attached device once confirmed it is reachable
+ if (index >= 0 && !mAvailableOutputDevices[index]->isAttached()) {
+ mAvailableOutputDevices[index]->attach(mHwModules[i]);
+ }
+ }
+ if (mPrimaryOutput == 0 &&
+ outProfile->mFlags & AUDIO_OUTPUT_FLAG_PRIMARY) {
+ mPrimaryOutput = output;
+ }
+ addOutput(output, outputDesc);
+ setOutputDevice(output,
+ outputDesc->mDevice,
+ true);
+ }
+ }
+ // open input streams needed to access attached devices to validate
+ // mAvailableInputDevices list
+ for (size_t j = 0; j < mHwModules[i]->mInputProfiles.size(); j++)
+ {
+ const sp<IOProfile> inProfile = mHwModules[i]->mInputProfiles[j];
+
+ if (inProfile->mSupportedDevices.isEmpty()) {
+ ALOGW("Input profile contains no device on module %s", mHwModules[i]->mName);
+ continue;
+ }
+ // chose first device present in mSupportedDevices also part of
+ // inputDeviceTypes
+ audio_devices_t profileType = AUDIO_DEVICE_NONE;
+ for (size_t k = 0; k < inProfile->mSupportedDevices.size(); k++) {
+ profileType = inProfile->mSupportedDevices[k]->mDeviceType;
+ if (profileType & inputDeviceTypes) {
+ break;
+ }
+ }
+ if ((profileType & inputDeviceTypes) == 0) {
+ continue;
+ }
+ sp<AudioInputDescriptor> inputDesc = new AudioInputDescriptor(inProfile);
+
+ inputDesc->mInputSource = AUDIO_SOURCE_MIC;
+ inputDesc->mDevice = profileType;
+
+ // find the address
+ DeviceVector inputDevices = mAvailableInputDevices.getDevicesFromType(profileType);
+ // the inputs vector must be of size 1, but we don't want to crash here
+ String8 address = inputDevices.size() > 0 ? inputDevices.itemAt(0)->mAddress
+ : String8("");
+ ALOGV(" for input device 0x%x using address %s", profileType, address.string());
+ ALOGE_IF(inputDevices.size() == 0, "Input device list is empty!");
+
+ audio_config_t config = AUDIO_CONFIG_INITIALIZER;
+ config.sample_rate = inputDesc->mSamplingRate;
+ config.channel_mask = inputDesc->mChannelMask;
+ config.format = inputDesc->mFormat;
+ audio_io_handle_t input = AUDIO_IO_HANDLE_NONE;
+ status_t status = mpClientInterface->openInput(inProfile->mModule->mHandle,
+ &input,
+ &config,
+ &inputDesc->mDevice,
+ address,
+ AUDIO_SOURCE_MIC,
+ AUDIO_INPUT_FLAG_NONE);
+
+ if (status == NO_ERROR) {
+ for (size_t k = 0; k < inProfile->mSupportedDevices.size(); k++) {
+ audio_devices_t type = inProfile->mSupportedDevices[k]->mDeviceType;
+ ssize_t index =
+ mAvailableInputDevices.indexOf(inProfile->mSupportedDevices[k]);
+ // give a valid ID to an attached device once confirmed it is reachable
+ if (index >= 0 && !mAvailableInputDevices[index]->isAttached()) {
+ mAvailableInputDevices[index]->attach(mHwModules[i]);
+ }
+ }
+ mpClientInterface->closeInput(input);
+ } else {
+ ALOGW("Cannot open input stream for device %08x on hw module %s",
+ inputDesc->mDevice,
+ mHwModules[i]->mName);
+ }
+ }
+ }
+ // make sure all attached devices have been allocated a unique ID
+ for (size_t i = 0; i < mAvailableOutputDevices.size();) {
+ if (!mAvailableOutputDevices[i]->isAttached()) {
+ ALOGW("Input device %08x unreachable", mAvailableOutputDevices[i]->mDeviceType);
+ mAvailableOutputDevices.remove(mAvailableOutputDevices[i]);
+ continue;
+ }
+ i++;
+ }
+ for (size_t i = 0; i < mAvailableInputDevices.size();) {
+ if (!mAvailableInputDevices[i]->isAttached()) {
+ ALOGW("Input device %08x unreachable", mAvailableInputDevices[i]->mDeviceType);
+ mAvailableInputDevices.remove(mAvailableInputDevices[i]);
+ continue;
+ }
+ i++;
+ }
+ // make sure default device is reachable
+ if (mAvailableOutputDevices.indexOf(mDefaultOutputDevice) < 0) {
+ ALOGE("Default device %08x is unreachable", mDefaultOutputDevice->mDeviceType);
+ }
+
+ ALOGE_IF((mPrimaryOutput == 0), "Failed to open primary output");
+
+ updateDevicesAndOutputs();
+
+#ifdef AUDIO_POLICY_TEST
+ if (mPrimaryOutput != 0) {
+ AudioParameter outputCmd = AudioParameter();
+ outputCmd.addInt(String8("set_id"), 0);
+ mpClientInterface->setParameters(mPrimaryOutput, outputCmd.toString());
+
+ mTestDevice = AUDIO_DEVICE_OUT_SPEAKER;
+ mTestSamplingRate = 44100;
+ mTestFormat = AUDIO_FORMAT_PCM_16_BIT;
+ mTestChannels = AUDIO_CHANNEL_OUT_STEREO;
+ mTestLatencyMs = 0;
+ mCurOutput = 0;
+ mDirectOutput = false;
+ for (int i = 0; i < NUM_TEST_OUTPUTS; i++) {
+ mTestOutputs[i] = 0;
+ }
+
+ const size_t SIZE = 256;
+ char buffer[SIZE];
+ snprintf(buffer, SIZE, "AudioPolicyManagerTest");
+ run(buffer, ANDROID_PRIORITY_AUDIO);
+ }
+#endif //AUDIO_POLICY_TEST
+}
+
+AudioPolicyManager::~AudioPolicyManager()
+{
+#ifdef AUDIO_POLICY_TEST
+ exit();
+#endif //AUDIO_POLICY_TEST
+ for (size_t i = 0; i < mOutputs.size(); i++) {
+ mpClientInterface->closeOutput(mOutputs.keyAt(i));
+ }
+ for (size_t i = 0; i < mInputs.size(); i++) {
+ mpClientInterface->closeInput(mInputs.keyAt(i));
+ }
+ mAvailableOutputDevices.clear();
+ mAvailableInputDevices.clear();
+ mOutputs.clear();
+ mInputs.clear();
+ mHwModules.clear();
+}
+
+status_t AudioPolicyManager::initCheck()
+{
+ return (mPrimaryOutput == 0) ? NO_INIT : NO_ERROR;
+}
+
+#ifdef AUDIO_POLICY_TEST
+bool AudioPolicyManager::threadLoop()
+{
+ ALOGV("entering threadLoop()");
+ while (!exitPending())
+ {
+ String8 command;
+ int valueInt;
+ String8 value;
+
+ Mutex::Autolock _l(mLock);
+ mWaitWorkCV.waitRelative(mLock, milliseconds(50));
+
+ command = mpClientInterface->getParameters(0, String8("test_cmd_policy"));
+ AudioParameter param = AudioParameter(command);
+
+ if (param.getInt(String8("test_cmd_policy"), valueInt) == NO_ERROR &&
+ valueInt != 0) {
+ ALOGV("Test command %s received", command.string());
+ String8 target;
+ if (param.get(String8("target"), target) != NO_ERROR) {
+ target = "Manager";
+ }
+ if (param.getInt(String8("test_cmd_policy_output"), valueInt) == NO_ERROR) {
+ param.remove(String8("test_cmd_policy_output"));
+ mCurOutput = valueInt;
+ }
+ if (param.get(String8("test_cmd_policy_direct"), value) == NO_ERROR) {
+ param.remove(String8("test_cmd_policy_direct"));
+ if (value == "false") {
+ mDirectOutput = false;
+ } else if (value == "true") {
+ mDirectOutput = true;
+ }
+ }
+ if (param.getInt(String8("test_cmd_policy_input"), valueInt) == NO_ERROR) {
+ param.remove(String8("test_cmd_policy_input"));
+ mTestInput = valueInt;
+ }
+
+ if (param.get(String8("test_cmd_policy_format"), value) == NO_ERROR) {
+ param.remove(String8("test_cmd_policy_format"));
+ int format = AUDIO_FORMAT_INVALID;
+ if (value == "PCM 16 bits") {
+ format = AUDIO_FORMAT_PCM_16_BIT;
+ } else if (value == "PCM 8 bits") {
+ format = AUDIO_FORMAT_PCM_8_BIT;
+ } else if (value == "Compressed MP3") {
+ format = AUDIO_FORMAT_MP3;
+ }
+ if (format != AUDIO_FORMAT_INVALID) {
+ if (target == "Manager") {
+ mTestFormat = format;
+ } else if (mTestOutputs[mCurOutput] != 0) {
+ AudioParameter outputParam = AudioParameter();
+ outputParam.addInt(String8("format"), format);
+ mpClientInterface->setParameters(mTestOutputs[mCurOutput], outputParam.toString());
+ }
+ }
+ }
+ if (param.get(String8("test_cmd_policy_channels"), value) == NO_ERROR) {
+ param.remove(String8("test_cmd_policy_channels"));
+ int channels = 0;
+
+ if (value == "Channels Stereo") {
+ channels = AUDIO_CHANNEL_OUT_STEREO;
+ } else if (value == "Channels Mono") {
+ channels = AUDIO_CHANNEL_OUT_MONO;
+ }
+ if (channels != 0) {
+ if (target == "Manager") {
+ mTestChannels = channels;
+ } else if (mTestOutputs[mCurOutput] != 0) {
+ AudioParameter outputParam = AudioParameter();
+ outputParam.addInt(String8("channels"), channels);
+ mpClientInterface->setParameters(mTestOutputs[mCurOutput], outputParam.toString());
+ }
+ }
+ }
+ if (param.getInt(String8("test_cmd_policy_sampleRate"), valueInt) == NO_ERROR) {
+ param.remove(String8("test_cmd_policy_sampleRate"));
+ if (valueInt >= 0 && valueInt <= 96000) {
+ int samplingRate = valueInt;
+ if (target == "Manager") {
+ mTestSamplingRate = samplingRate;
+ } else if (mTestOutputs[mCurOutput] != 0) {
+ AudioParameter outputParam = AudioParameter();
+ outputParam.addInt(String8("sampling_rate"), samplingRate);
+ mpClientInterface->setParameters(mTestOutputs[mCurOutput], outputParam.toString());
+ }
+ }
+ }
+
+ if (param.get(String8("test_cmd_policy_reopen"), value) == NO_ERROR) {
+ param.remove(String8("test_cmd_policy_reopen"));
+
+ sp<AudioOutputDescriptor> outputDesc = mOutputs.valueFor(mPrimaryOutput);
+ mpClientInterface->closeOutput(mPrimaryOutput);
+
+ audio_module_handle_t moduleHandle = outputDesc->mModule->mHandle;
+
+ mOutputs.removeItem(mPrimaryOutput);
+
+ sp<AudioOutputDescriptor> outputDesc = new AudioOutputDescriptor(NULL);
+ outputDesc->mDevice = AUDIO_DEVICE_OUT_SPEAKER;
+ audio_config_t config = AUDIO_CONFIG_INITIALIZER;
+ config.sample_rate = outputDesc->mSamplingRate;
+ config.channel_mask = outputDesc->mChannelMask;
+ config.format = outputDesc->mFormat;
+ status_t status = mpClientInterface->openOutput(moduleHandle,
+ &mPrimaryOutput,
+ &config,
+ &outputDesc->mDevice,
+ String8(""),
+ &outputDesc->mLatency,
+ outputDesc->mFlags);
+ if (status != NO_ERROR) {
+ ALOGE("Failed to reopen hardware output stream, "
+ "samplingRate: %d, format %d, channels %d",
+ outputDesc->mSamplingRate, outputDesc->mFormat, outputDesc->mChannelMask);
+ } else {
+ outputDesc->mSamplingRate = config.sample_rate;
+ outputDesc->mChannelMask = config.channel_mask;
+ outputDesc->mFormat = config.format;
+ AudioParameter outputCmd = AudioParameter();
+ outputCmd.addInt(String8("set_id"), 0);
+ mpClientInterface->setParameters(mPrimaryOutput, outputCmd.toString());
+ addOutput(mPrimaryOutput, outputDesc);
+ }
+ }
+
+
+ mpClientInterface->setParameters(0, String8("test_cmd_policy="));
+ }
+ }
+ return false;
+}
+
+void AudioPolicyManager::exit()
+{
+ {
+ AutoMutex _l(mLock);
+ requestExit();
+ mWaitWorkCV.signal();
+ }
+ requestExitAndWait();
+}
+
+int AudioPolicyManager::testOutputIndex(audio_io_handle_t output)
+{
+ for (int i = 0; i < NUM_TEST_OUTPUTS; i++) {
+ if (output == mTestOutputs[i]) return i;
+ }
+ return 0;
+}
+#endif //AUDIO_POLICY_TEST
+
+// ---
+
+void AudioPolicyManager::addOutput(audio_io_handle_t output, sp<AudioOutputDescriptor> outputDesc)
+{
+ outputDesc->mIoHandle = output;
+ outputDesc->mId = nextUniqueId();
+ mOutputs.add(output, outputDesc);
+ nextAudioPortGeneration();
+}
+
+void AudioPolicyManager::addInput(audio_io_handle_t input, sp<AudioInputDescriptor> inputDesc)
+{
+ inputDesc->mIoHandle = input;
+ inputDesc->mId = nextUniqueId();
+ mInputs.add(input, inputDesc);
+ nextAudioPortGeneration();
+}
+
+void AudioPolicyManager::findIoHandlesByAddress(sp<AudioOutputDescriptor> desc /*in*/,
+ const audio_devices_t device /*in*/,
+ const String8 address /*in*/,
+ SortedVector<audio_io_handle_t>& outputs /*out*/) {
+ sp<DeviceDescriptor> devDesc =
+ desc->mProfile->mSupportedDevices.getDevice(device, address);
+ if (devDesc != 0) {
+ ALOGV("findIoHandlesByAddress(): adding opened output %d on same address %s",
+ desc->mIoHandle, address.string());
+ outputs.add(desc->mIoHandle);
+ }
+}
+
+status_t AudioPolicyManager::checkOutputsForDevice(const sp<DeviceDescriptor> devDesc,
+ audio_policy_dev_state_t state,
+ SortedVector<audio_io_handle_t>& outputs,
+ const String8 address)
+{
+ audio_devices_t device = devDesc->mDeviceType;
+ sp<AudioOutputDescriptor> desc;
+ // erase all current sample rates, formats and channel masks
+ devDesc->clearCapabilities();
+
+ if (state == AUDIO_POLICY_DEVICE_STATE_AVAILABLE) {
+ // first list already open outputs that can be routed to this device
+ for (size_t i = 0; i < mOutputs.size(); i++) {
+ desc = mOutputs.valueAt(i);
+ if (!desc->isDuplicated() && (desc->mProfile->mSupportedDevices.types() & device)) {
+ if (!deviceDistinguishesOnAddress(device)) {
+ ALOGV("checkOutputsForDevice(): adding opened output %d", mOutputs.keyAt(i));
+ outputs.add(mOutputs.keyAt(i));
+ } else {
+ ALOGV(" checking address match due to device 0x%x", device);
+ findIoHandlesByAddress(desc, device, address, outputs);
+ }
+ }
+ }
+ // then look for output profiles that can be routed to this device
+ SortedVector< sp<IOProfile> > profiles;
+ for (size_t i = 0; i < mHwModules.size(); i++)
+ {
+ if (mHwModules[i]->mHandle == 0) {
+ continue;
+ }
+ for (size_t j = 0; j < mHwModules[i]->mOutputProfiles.size(); j++)
+ {
+ sp<IOProfile> profile = mHwModules[i]->mOutputProfiles[j];
+ if (profile->mSupportedDevices.types() & device) {
+ if (!deviceDistinguishesOnAddress(device) ||
+ address == profile->mSupportedDevices[0]->mAddress) {
+ profiles.add(profile);
+ ALOGV("checkOutputsForDevice(): adding profile %zu from module %zu", j, i);
+ }
+ }
+ }
+ }
+
+ ALOGV(" found %d profiles, %d outputs", profiles.size(), outputs.size());
+
+ if (profiles.isEmpty() && outputs.isEmpty()) {
+ ALOGW("checkOutputsForDevice(): No output available for device %04x", device);
+ return BAD_VALUE;
+ }
+
+ // open outputs for matching profiles if needed. Direct outputs are also opened to
+ // query for dynamic parameters and will be closed later by setDeviceConnectionState()
+ for (ssize_t profile_index = 0; profile_index < (ssize_t)profiles.size(); profile_index++) {
+ sp<IOProfile> profile = profiles[profile_index];
+
+ // nothing to do if one output is already opened for this profile
+ size_t j;
+ for (j = 0; j < outputs.size(); j++) {
+ desc = mOutputs.valueFor(outputs.itemAt(j));
+ if (!desc->isDuplicated() && desc->mProfile == profile) {
+ // matching profile: save the sample rates, format and channel masks supported
+ // by the profile in our device descriptor
+ devDesc->importAudioPort(profile);
+ break;
+ }
+ }
+ if (j != outputs.size()) {
+ continue;
+ }
+
+ ALOGV("opening output for device %08x with params %s profile %p",
+ device, address.string(), profile.get());
+ desc = new AudioOutputDescriptor(profile);
+ desc->mDevice = device;
+ audio_config_t config = AUDIO_CONFIG_INITIALIZER;
+ config.sample_rate = desc->mSamplingRate;
+ config.channel_mask = desc->mChannelMask;
+ config.format = desc->mFormat;
+ config.offload_info.sample_rate = desc->mSamplingRate;
+ config.offload_info.channel_mask = desc->mChannelMask;
+ config.offload_info.format = desc->mFormat;
+ audio_io_handle_t output = AUDIO_IO_HANDLE_NONE;
+ status_t status = mpClientInterface->openOutput(profile->mModule->mHandle,
+ &output,
+ &config,
+ &desc->mDevice,
+ address,
+ &desc->mLatency,
+ desc->mFlags);
+ if (status == NO_ERROR) {
+ desc->mSamplingRate = config.sample_rate;
+ desc->mChannelMask = config.channel_mask;
+ desc->mFormat = config.format;
+
+ // Here is where the out_set_parameters() for card & device gets called
+ if (!address.isEmpty()) {
+ char *param = audio_device_address_to_parameter(device, address);
+ mpClientInterface->setParameters(output, String8(param));
+ free(param);
+ }
+
+ // Here is where we step through and resolve any "dynamic" fields
+ String8 reply;
+ char *value;
+ if (profile->mSamplingRates[0] == 0) {
+ reply = mpClientInterface->getParameters(output,
+ String8(AUDIO_PARAMETER_STREAM_SUP_SAMPLING_RATES));
+ ALOGV("checkOutputsForDevice() supported sampling rates %s",
+ reply.string());
+ value = strpbrk((char *)reply.string(), "=");
+ if (value != NULL) {
+ profile->loadSamplingRates(value + 1);
+ }
+ }
+ if (profile->mFormats[0] == AUDIO_FORMAT_DEFAULT) {
+ reply = mpClientInterface->getParameters(output,
+ String8(AUDIO_PARAMETER_STREAM_SUP_FORMATS));
+ ALOGV("checkOutputsForDevice() supported formats %s",
+ reply.string());
+ value = strpbrk((char *)reply.string(), "=");
+ if (value != NULL) {
+ profile->loadFormats(value + 1);
+ }
+ }
+ if (profile->mChannelMasks[0] == 0) {
+ reply = mpClientInterface->getParameters(output,
+ String8(AUDIO_PARAMETER_STREAM_SUP_CHANNELS));
+ ALOGV("checkOutputsForDevice() supported channel masks %s",
+ reply.string());
+ value = strpbrk((char *)reply.string(), "=");
+ if (value != NULL) {
+ profile->loadOutChannels(value + 1);
+ }
+ }
+ if (((profile->mSamplingRates[0] == 0) &&
+ (profile->mSamplingRates.size() < 2)) ||
+ ((profile->mFormats[0] == AUDIO_FORMAT_DEFAULT) &&
+ (profile->mFormats.size() < 2)) ||
+ ((profile->mChannelMasks[0] == 0) &&
+ (profile->mChannelMasks.size() < 2))) {
+ ALOGW("checkOutputsForDevice() missing param");
+ mpClientInterface->closeOutput(output);
+ output = AUDIO_IO_HANDLE_NONE;
+ } else if (profile->mSamplingRates[0] == 0 || profile->mFormats[0] == 0 ||
+ profile->mChannelMasks[0] == 0) {
+ mpClientInterface->closeOutput(output);
+ config.sample_rate = profile->pickSamplingRate();
+ config.channel_mask = profile->pickChannelMask();
+ config.format = profile->pickFormat();
+ config.offload_info.sample_rate = config.sample_rate;
+ config.offload_info.channel_mask = config.channel_mask;
+ config.offload_info.format = config.format;
+ status = mpClientInterface->openOutput(profile->mModule->mHandle,
+ &output,
+ &config,
+ &desc->mDevice,
+ address,
+ &desc->mLatency,
+ desc->mFlags);
+ if (status == NO_ERROR) {
+ desc->mSamplingRate = config.sample_rate;
+ desc->mChannelMask = config.channel_mask;
+ desc->mFormat = config.format;
+ } else {
+ output = AUDIO_IO_HANDLE_NONE;
+ }
+ }
+
+ if (output != AUDIO_IO_HANDLE_NONE) {
+ addOutput(output, desc);
+ if (deviceDistinguishesOnAddress(device) && address != "0") {
+ ssize_t index = mPolicyMixes.indexOfKey(address);
+ if (index >= 0) {
+ mPolicyMixes[index]->mOutput = desc;
+ desc->mPolicyMix = &mPolicyMixes[index]->mMix;
+ } else {
+ ALOGE("checkOutputsForDevice() cannot find policy for address %s",
+ address.string());
+ }
+ } else if ((desc->mFlags & AUDIO_OUTPUT_FLAG_DIRECT) == 0) {
+ // no duplicated output for direct outputs and
+ // outputs used by dynamic policy mixes
+ audio_io_handle_t duplicatedOutput = AUDIO_IO_HANDLE_NONE;
+
+ // set initial stream volume for device
+ applyStreamVolumes(output, device, 0, true);
+
+ //TODO: configure audio effect output stage here
+
+ // open a duplicating output thread for the new output and the primary output
+ duplicatedOutput = mpClientInterface->openDuplicateOutput(output,
+ mPrimaryOutput);
+ if (duplicatedOutput != AUDIO_IO_HANDLE_NONE) {
+ // add duplicated output descriptor
+ sp<AudioOutputDescriptor> dupOutputDesc =
+ new AudioOutputDescriptor(NULL);
+ dupOutputDesc->mOutput1 = mOutputs.valueFor(mPrimaryOutput);
+ dupOutputDesc->mOutput2 = mOutputs.valueFor(output);
+ dupOutputDesc->mSamplingRate = desc->mSamplingRate;
+ dupOutputDesc->mFormat = desc->mFormat;
+ dupOutputDesc->mChannelMask = desc->mChannelMask;
+ dupOutputDesc->mLatency = desc->mLatency;
+ addOutput(duplicatedOutput, dupOutputDesc);
+ applyStreamVolumes(duplicatedOutput, device, 0, true);
+ } else {
+ ALOGW("checkOutputsForDevice() could not open dup output for %d and %d",
+ mPrimaryOutput, output);
+ mpClientInterface->closeOutput(output);
+ mOutputs.removeItem(output);
+ nextAudioPortGeneration();
+ output = AUDIO_IO_HANDLE_NONE;
+ }
+ }
+ }
+ } else {
+ output = AUDIO_IO_HANDLE_NONE;
+ }
+ if (output == AUDIO_IO_HANDLE_NONE) {
+ ALOGW("checkOutputsForDevice() could not open output for device %x", device);
+ profiles.removeAt(profile_index);
+ profile_index--;
+ } else {
+ outputs.add(output);
+ devDesc->importAudioPort(profile);
+
+ if (deviceDistinguishesOnAddress(device)) {
+ ALOGV("checkOutputsForDevice(): setOutputDevice(dev=0x%x, addr=%s)",
+ device, address.string());
+ setOutputDevice(output, device, true/*force*/, 0/*delay*/,
+ NULL/*patch handle*/, address.string());
+ }
+ ALOGV("checkOutputsForDevice(): adding output %d", output);
+ }
+ }
+
+ if (profiles.isEmpty()) {
+ ALOGW("checkOutputsForDevice(): No output available for device %04x", device);
+ return BAD_VALUE;
+ }
+ } else { // Disconnect
+ // check if one opened output is not needed any more after disconnecting one device
+ for (size_t i = 0; i < mOutputs.size(); i++) {
+ desc = mOutputs.valueAt(i);
+ if (!desc->isDuplicated()) {
+ // exact match on device
+ if (deviceDistinguishesOnAddress(device) &&
+ (desc->mProfile->mSupportedDevices.types() == device)) {
+ findIoHandlesByAddress(desc, device, address, outputs);
+ } else if (!(desc->mProfile->mSupportedDevices.types()
+ & mAvailableOutputDevices.types())) {
+ ALOGV("checkOutputsForDevice(): disconnecting adding output %d",
+ mOutputs.keyAt(i));
+ outputs.add(mOutputs.keyAt(i));
+ }
+ }
+ }
+ // Clear any profiles associated with the disconnected device.
+ for (size_t i = 0; i < mHwModules.size(); i++)
+ {
+ if (mHwModules[i]->mHandle == 0) {
+ continue;
+ }
+ for (size_t j = 0; j < mHwModules[i]->mOutputProfiles.size(); j++)
+ {
+ sp<IOProfile> profile = mHwModules[i]->mOutputProfiles[j];
+ if (profile->mSupportedDevices.types() & device) {
+ ALOGV("checkOutputsForDevice(): "
+ "clearing direct output profile %zu on module %zu", j, i);
+ if (profile->mSamplingRates[0] == 0) {
+ profile->mSamplingRates.clear();
+ profile->mSamplingRates.add(0);
+ }
+ if (profile->mFormats[0] == AUDIO_FORMAT_DEFAULT) {
+ profile->mFormats.clear();
+ profile->mFormats.add(AUDIO_FORMAT_DEFAULT);
+ }
+ if (profile->mChannelMasks[0] == 0) {
+ profile->mChannelMasks.clear();
+ profile->mChannelMasks.add(0);
+ }
+ }
+ }
+ }
+ }
+ return NO_ERROR;
+}
+
+status_t AudioPolicyManager::checkInputsForDevice(audio_devices_t device,
+ audio_policy_dev_state_t state,
+ SortedVector<audio_io_handle_t>& inputs,
+ const String8 address)
+{
+ sp<AudioInputDescriptor> desc;
+ if (state == AUDIO_POLICY_DEVICE_STATE_AVAILABLE) {
+ // first list already open inputs that can be routed to this device
+ for (size_t input_index = 0; input_index < mInputs.size(); input_index++) {
+ desc = mInputs.valueAt(input_index);
+ if (desc->mProfile->mSupportedDevices.types() & (device & ~AUDIO_DEVICE_BIT_IN)) {
+ ALOGV("checkInputsForDevice(): adding opened input %d", mInputs.keyAt(input_index));
+ inputs.add(mInputs.keyAt(input_index));
+ }
+ }
+
+ // then look for input profiles that can be routed to this device
+ SortedVector< sp<IOProfile> > profiles;
+ for (size_t module_idx = 0; module_idx < mHwModules.size(); module_idx++)
+ {
+ if (mHwModules[module_idx]->mHandle == 0) {
+ continue;
+ }
+ for (size_t profile_index = 0;
+ profile_index < mHwModules[module_idx]->mInputProfiles.size();
+ profile_index++)
+ {
+ sp<IOProfile> profile = mHwModules[module_idx]->mInputProfiles[profile_index];
+
+ if (profile->mSupportedDevices.types() & (device & ~AUDIO_DEVICE_BIT_IN)) {
+ if (!deviceDistinguishesOnAddress(device) ||
+ address == profile->mSupportedDevices[0]->mAddress) {
+ profiles.add(profile);
+ ALOGV("checkInputsForDevice(): adding profile %zu from module %zu",
+ profile_index, module_idx);
+ }
+ }
+ }
+ }
+
+ if (profiles.isEmpty() && inputs.isEmpty()) {
+ ALOGW("checkInputsForDevice(): No input available for device 0x%X", device);
+ return BAD_VALUE;
+ }
+
+ // open inputs for matching profiles if needed. Direct inputs are also opened to
+ // query for dynamic parameters and will be closed later by setDeviceConnectionState()
+ for (ssize_t profile_index = 0; profile_index < (ssize_t)profiles.size(); profile_index++) {
+
+ sp<IOProfile> profile = profiles[profile_index];
+ // nothing to do if one input is already opened for this profile
+ size_t input_index;
+ for (input_index = 0; input_index < mInputs.size(); input_index++) {
+ desc = mInputs.valueAt(input_index);
+ if (desc->mProfile == profile) {
+ break;
+ }
+ }
+ if (input_index != mInputs.size()) {
+ continue;
+ }
+
+ ALOGV("opening input for device 0x%X with params %s", device, address.string());
+ desc = new AudioInputDescriptor(profile);
+ desc->mDevice = device;
+ audio_config_t config = AUDIO_CONFIG_INITIALIZER;
+ config.sample_rate = desc->mSamplingRate;
+ config.channel_mask = desc->mChannelMask;
+ config.format = desc->mFormat;
+ audio_io_handle_t input = AUDIO_IO_HANDLE_NONE;
+ status_t status = mpClientInterface->openInput(profile->mModule->mHandle,
+ &input,
+ &config,
+ &desc->mDevice,
+ address,
+ AUDIO_SOURCE_MIC,
+ AUDIO_INPUT_FLAG_NONE /*FIXME*/);
+
+ if (status == NO_ERROR) {
+ desc->mSamplingRate = config.sample_rate;
+ desc->mChannelMask = config.channel_mask;
+ desc->mFormat = config.format;
+
+ if (!address.isEmpty()) {
+ char *param = audio_device_address_to_parameter(device, address);
+ mpClientInterface->setParameters(input, String8(param));
+ free(param);
+ }
+
+ // Here is where we step through and resolve any "dynamic" fields
+ String8 reply;
+ char *value;
+ if (profile->mSamplingRates[0] == 0) {
+ reply = mpClientInterface->getParameters(input,
+ String8(AUDIO_PARAMETER_STREAM_SUP_SAMPLING_RATES));
+ ALOGV("checkInputsForDevice() direct input sup sampling rates %s",
+ reply.string());
+ value = strpbrk((char *)reply.string(), "=");
+ if (value != NULL) {
+ profile->loadSamplingRates(value + 1);
+ }
+ }
+ if (profile->mFormats[0] == AUDIO_FORMAT_DEFAULT) {
+ reply = mpClientInterface->getParameters(input,
+ String8(AUDIO_PARAMETER_STREAM_SUP_FORMATS));
+ ALOGV("checkInputsForDevice() direct input sup formats %s", reply.string());
+ value = strpbrk((char *)reply.string(), "=");
+ if (value != NULL) {
+ profile->loadFormats(value + 1);
+ }
+ }
+ if (profile->mChannelMasks[0] == 0) {
+ reply = mpClientInterface->getParameters(input,
+ String8(AUDIO_PARAMETER_STREAM_SUP_CHANNELS));
+ ALOGV("checkInputsForDevice() direct input sup channel masks %s",
+ reply.string());
+ value = strpbrk((char *)reply.string(), "=");
+ if (value != NULL) {
+ profile->loadInChannels(value + 1);
+ }
+ }
+ if (((profile->mSamplingRates[0] == 0) && (profile->mSamplingRates.size() < 2)) ||
+ ((profile->mFormats[0] == 0) && (profile->mFormats.size() < 2)) ||
+ ((profile->mChannelMasks[0] == 0) && (profile->mChannelMasks.size() < 2))) {
+ ALOGW("checkInputsForDevice() direct input missing param");
+ mpClientInterface->closeInput(input);
+ input = AUDIO_IO_HANDLE_NONE;
+ }
+
+ if (input != 0) {
+ addInput(input, desc);
+ }
+ } // endif input != 0
+
+ if (input == AUDIO_IO_HANDLE_NONE) {
+ ALOGW("checkInputsForDevice() could not open input for device 0x%X", device);
+ profiles.removeAt(profile_index);
+ profile_index--;
+ } else {
+ inputs.add(input);
+ ALOGV("checkInputsForDevice(): adding input %d", input);
+ }
+ } // end scan profiles
+
+ if (profiles.isEmpty()) {
+ ALOGW("checkInputsForDevice(): No input available for device 0x%X", device);
+ return BAD_VALUE;
+ }
+ } else {
+ // Disconnect
+ // check if one opened input is not needed any more after disconnecting one device
+ for (size_t input_index = 0; input_index < mInputs.size(); input_index++) {
+ desc = mInputs.valueAt(input_index);
+ if (!(desc->mProfile->mSupportedDevices.types() & mAvailableInputDevices.types() &
+ ~AUDIO_DEVICE_BIT_IN)) {
+ ALOGV("checkInputsForDevice(): disconnecting adding input %d",
+ mInputs.keyAt(input_index));
+ inputs.add(mInputs.keyAt(input_index));
+ }
+ }
+ // Clear any profiles associated with the disconnected device.
+ for (size_t module_index = 0; module_index < mHwModules.size(); module_index++) {
+ if (mHwModules[module_index]->mHandle == 0) {
+ continue;
+ }
+ for (size_t profile_index = 0;
+ profile_index < mHwModules[module_index]->mInputProfiles.size();
+ profile_index++) {
+ sp<IOProfile> profile = mHwModules[module_index]->mInputProfiles[profile_index];
+ if (profile->mSupportedDevices.types() & device & ~AUDIO_DEVICE_BIT_IN) {
+ ALOGV("checkInputsForDevice(): clearing direct input profile %zu on module %zu",
+ profile_index, module_index);
+ if (profile->mSamplingRates[0] == 0) {
+ profile->mSamplingRates.clear();
+ profile->mSamplingRates.add(0);
+ }
+ if (profile->mFormats[0] == AUDIO_FORMAT_DEFAULT) {
+ profile->mFormats.clear();
+ profile->mFormats.add(AUDIO_FORMAT_DEFAULT);
+ }
+ if (profile->mChannelMasks[0] == 0) {
+ profile->mChannelMasks.clear();
+ profile->mChannelMasks.add(0);
+ }
+ }
+ }
+ }
+ } // end disconnect
+
+ return NO_ERROR;
+}
+
+
+void AudioPolicyManager::closeOutput(audio_io_handle_t output)
+{
+ ALOGV("closeOutput(%d)", output);
+
+ sp<AudioOutputDescriptor> outputDesc = mOutputs.valueFor(output);
+ if (outputDesc == NULL) {
+ ALOGW("closeOutput() unknown output %d", output);
+ return;
+ }
+
+ for (size_t i = 0; i < mPolicyMixes.size(); i++) {
+ if (mPolicyMixes[i]->mOutput == outputDesc) {
+ mPolicyMixes[i]->mOutput.clear();
+ }
+ }
+
+ // look for duplicated outputs connected to the output being removed.
+ for (size_t i = 0; i < mOutputs.size(); i++) {
+ sp<AudioOutputDescriptor> dupOutputDesc = mOutputs.valueAt(i);
+ if (dupOutputDesc->isDuplicated() &&
+ (dupOutputDesc->mOutput1 == outputDesc ||
+ dupOutputDesc->mOutput2 == outputDesc)) {
+ sp<AudioOutputDescriptor> outputDesc2;
+ if (dupOutputDesc->mOutput1 == outputDesc) {
+ outputDesc2 = dupOutputDesc->mOutput2;
+ } else {
+ outputDesc2 = dupOutputDesc->mOutput1;
+ }
+ // As all active tracks on duplicated output will be deleted,
+ // and as they were also referenced on the other output, the reference
+ // count for their stream type must be adjusted accordingly on
+ // the other output.
+ for (int j = 0; j < AUDIO_STREAM_CNT; j++) {
+ int refCount = dupOutputDesc->mRefCount[j];
+ outputDesc2->changeRefCount((audio_stream_type_t)j,-refCount);
+ }
+ audio_io_handle_t duplicatedOutput = mOutputs.keyAt(i);
+ ALOGV("closeOutput() closing also duplicated output %d", duplicatedOutput);
+
+ mpClientInterface->closeOutput(duplicatedOutput);
+ mOutputs.removeItem(duplicatedOutput);
+ }
+ }
+
+ nextAudioPortGeneration();
+
+ ssize_t index = mAudioPatches.indexOfKey(outputDesc->mPatchHandle);
+ if (index >= 0) {
+ sp<AudioPatch> patchDesc = mAudioPatches.valueAt(index);
+ status_t status = mpClientInterface->releaseAudioPatch(patchDesc->mAfPatchHandle, 0);
+ mAudioPatches.removeItemsAt(index);
+ mpClientInterface->onAudioPatchListUpdate();
+ }
+
+ AudioParameter param;
+ param.add(String8("closing"), String8("true"));
+ mpClientInterface->setParameters(output, param.toString());
+
+ mpClientInterface->closeOutput(output);
+ mOutputs.removeItem(output);
+ mPreviousOutputs = mOutputs;
+}
+
+void AudioPolicyManager::closeInput(audio_io_handle_t input)
+{
+ ALOGV("closeInput(%d)", input);
+
+ sp<AudioInputDescriptor> inputDesc = mInputs.valueFor(input);
+ if (inputDesc == NULL) {
+ ALOGW("closeInput() unknown input %d", input);
+ return;
+ }
+
+ nextAudioPortGeneration();
+
+ ssize_t index = mAudioPatches.indexOfKey(inputDesc->mPatchHandle);
+ if (index >= 0) {
+ sp<AudioPatch> patchDesc = mAudioPatches.valueAt(index);
+ status_t status = mpClientInterface->releaseAudioPatch(patchDesc->mAfPatchHandle, 0);
+ mAudioPatches.removeItemsAt(index);
+ mpClientInterface->onAudioPatchListUpdate();
+ }
+
+ mpClientInterface->closeInput(input);
+ mInputs.removeItem(input);
+}
+
+SortedVector<audio_io_handle_t> AudioPolicyManager::getOutputsForDevice(audio_devices_t device,
+ DefaultKeyedVector<audio_io_handle_t, sp<AudioOutputDescriptor> > openOutputs)
+{
+ SortedVector<audio_io_handle_t> outputs;
+
+ ALOGVV("getOutputsForDevice() device %04x", device);
+ for (size_t i = 0; i < openOutputs.size(); i++) {
+ ALOGVV("output %d isDuplicated=%d device=%04x",
+ i, openOutputs.valueAt(i)->isDuplicated(), openOutputs.valueAt(i)->supportedDevices());
+ if ((device & openOutputs.valueAt(i)->supportedDevices()) == device) {
+ ALOGVV("getOutputsForDevice() found output %d", openOutputs.keyAt(i));
+ outputs.add(openOutputs.keyAt(i));
+ }
+ }
+ return outputs;
+}
+
+bool AudioPolicyManager::vectorsEqual(SortedVector<audio_io_handle_t>& outputs1,
+ SortedVector<audio_io_handle_t>& outputs2)
+{
+ if (outputs1.size() != outputs2.size()) {
+ return false;
+ }
+ for (size_t i = 0; i < outputs1.size(); i++) {
+ if (outputs1[i] != outputs2[i]) {
+ return false;
+ }
+ }
+ return true;
+}
+
+void AudioPolicyManager::checkOutputForStrategy(routing_strategy strategy)
+{
+ audio_devices_t oldDevice = getDeviceForStrategy(strategy, true /*fromCache*/);
+ audio_devices_t newDevice = getDeviceForStrategy(strategy, false /*fromCache*/);
+ SortedVector<audio_io_handle_t> srcOutputs = getOutputsForDevice(oldDevice, mPreviousOutputs);
+ SortedVector<audio_io_handle_t> dstOutputs = getOutputsForDevice(newDevice, mOutputs);
+
+ // also take into account external policy-related changes: add all outputs which are
+ // associated with policies in the "before" and "after" output vectors
+ ALOGVV("checkOutputForStrategy(): policy related outputs");
+ for (size_t i = 0 ; i < mPreviousOutputs.size() ; i++) {
+ const sp<AudioOutputDescriptor> desc = mPreviousOutputs.valueAt(i);
+ if (desc != 0 && desc->mPolicyMix != NULL) {
+ srcOutputs.add(desc->mIoHandle);
+ ALOGVV(" previous outputs: adding %d", desc->mIoHandle);
+ }
+ }
+ for (size_t i = 0 ; i < mOutputs.size() ; i++) {
+ const sp<AudioOutputDescriptor> desc = mOutputs.valueAt(i);
+ if (desc != 0 && desc->mPolicyMix != NULL) {
+ dstOutputs.add(desc->mIoHandle);
+ ALOGVV(" new outputs: adding %d", desc->mIoHandle);
+ }
+ }
+
+ if (!vectorsEqual(srcOutputs,dstOutputs)) {
+ ALOGV("checkOutputForStrategy() strategy %d, moving from output %d to output %d",
+ strategy, srcOutputs[0], dstOutputs[0]);
+ // mute strategy while moving tracks from one output to another
+ for (size_t i = 0; i < srcOutputs.size(); i++) {
+ sp<AudioOutputDescriptor> desc = mOutputs.valueFor(srcOutputs[i]);
+ if (desc->isStrategyActive(strategy)) {
+ setStrategyMute(strategy, true, srcOutputs[i]);
+ setStrategyMute(strategy, false, srcOutputs[i], MUTE_TIME_MS, newDevice);
+ }
+ }
+
+ // Move effects associated to this strategy from previous output to new output
+ if (strategy == STRATEGY_MEDIA) {
+ audio_io_handle_t fxOutput = selectOutputForEffects(dstOutputs);
+ SortedVector<audio_io_handle_t> moved;
+ for (size_t i = 0; i < mEffects.size(); i++) {
+ sp<EffectDescriptor> effectDesc = mEffects.valueAt(i);
+ if (effectDesc->mSession == AUDIO_SESSION_OUTPUT_MIX &&
+ effectDesc->mIo != fxOutput) {
+ if (moved.indexOf(effectDesc->mIo) < 0) {
+ ALOGV("checkOutputForStrategy() moving effect %d to output %d",
+ mEffects.keyAt(i), fxOutput);
+ mpClientInterface->moveEffects(AUDIO_SESSION_OUTPUT_MIX, effectDesc->mIo,
+ fxOutput);
+ moved.add(effectDesc->mIo);
+ }
+ effectDesc->mIo = fxOutput;
+ }
+ }
+ }
+ // Move tracks associated to this strategy from previous output to new output
+ for (int i = 0; i < AUDIO_STREAM_CNT; i++) {
+ if (i == AUDIO_STREAM_PATCH) {
+ continue;
+ }
+ if (getStrategy((audio_stream_type_t)i) == strategy) {
+ mpClientInterface->invalidateStream((audio_stream_type_t)i);
+ }
+ }
+ }
+}
+
+void AudioPolicyManager::checkOutputForAllStrategies()
+{
+ if (mForceUse[AUDIO_POLICY_FORCE_FOR_SYSTEM] == AUDIO_POLICY_FORCE_SYSTEM_ENFORCED)
+ checkOutputForStrategy(STRATEGY_ENFORCED_AUDIBLE);
+ checkOutputForStrategy(STRATEGY_PHONE);
+ if (mForceUse[AUDIO_POLICY_FORCE_FOR_SYSTEM] != AUDIO_POLICY_FORCE_SYSTEM_ENFORCED)
+ checkOutputForStrategy(STRATEGY_ENFORCED_AUDIBLE);
+ checkOutputForStrategy(STRATEGY_SONIFICATION);
+ checkOutputForStrategy(STRATEGY_SONIFICATION_RESPECTFUL);
+ checkOutputForStrategy(STRATEGY_ACCESSIBILITY);
+ checkOutputForStrategy(STRATEGY_MEDIA);
+ checkOutputForStrategy(STRATEGY_DTMF);
+ checkOutputForStrategy(STRATEGY_REROUTING);
+}
+
+audio_io_handle_t AudioPolicyManager::getA2dpOutput()
+{
+ for (size_t i = 0; i < mOutputs.size(); i++) {
+ sp<AudioOutputDescriptor> outputDesc = mOutputs.valueAt(i);
+ if (!outputDesc->isDuplicated() && outputDesc->device() & AUDIO_DEVICE_OUT_ALL_A2DP) {
+ return mOutputs.keyAt(i);
+ }
+ }
+
+ return 0;
+}
+
+void AudioPolicyManager::checkA2dpSuspend()
+{
+ audio_io_handle_t a2dpOutput = getA2dpOutput();
+ if (a2dpOutput == 0) {
+ mA2dpSuspended = false;
+ return;
+ }
+
+ bool isScoConnected =
+ ((mAvailableInputDevices.types() & AUDIO_DEVICE_IN_BLUETOOTH_SCO_HEADSET &
+ ~AUDIO_DEVICE_BIT_IN) != 0) ||
+ ((mAvailableOutputDevices.types() & AUDIO_DEVICE_OUT_ALL_SCO) != 0);
+ // suspend A2DP output if:
+ // (NOT already suspended) &&
+ // ((SCO device is connected &&
+ // (forced usage for communication || for record is SCO))) ||
+ // (phone state is ringing || in call)
+ //
+ // restore A2DP output if:
+ // (Already suspended) &&
+ // ((SCO device is NOT connected ||
+ // (forced usage NOT for communication && NOT for record is SCO))) &&
+ // (phone state is NOT ringing && NOT in call)
+ //
+ if (mA2dpSuspended) {
+ if ((!isScoConnected ||
+ ((mForceUse[AUDIO_POLICY_FORCE_FOR_COMMUNICATION] != AUDIO_POLICY_FORCE_BT_SCO) &&
+ (mForceUse[AUDIO_POLICY_FORCE_FOR_RECORD] != AUDIO_POLICY_FORCE_BT_SCO))) &&
+ ((mPhoneState != AUDIO_MODE_IN_CALL) &&
+ (mPhoneState != AUDIO_MODE_RINGTONE))) {
+
+ mpClientInterface->restoreOutput(a2dpOutput);
+ mA2dpSuspended = false;
+ }
+ } else {
+ if ((isScoConnected &&
+ ((mForceUse[AUDIO_POLICY_FORCE_FOR_COMMUNICATION] == AUDIO_POLICY_FORCE_BT_SCO) ||
+ (mForceUse[AUDIO_POLICY_FORCE_FOR_RECORD] == AUDIO_POLICY_FORCE_BT_SCO))) ||
+ ((mPhoneState == AUDIO_MODE_IN_CALL) ||
+ (mPhoneState == AUDIO_MODE_RINGTONE))) {
+
+ mpClientInterface->suspendOutput(a2dpOutput);
+ mA2dpSuspended = true;
+ }
+ }
+}
+
+audio_devices_t AudioPolicyManager::getNewOutputDevice(audio_io_handle_t output, bool fromCache)
+{
+ audio_devices_t device = AUDIO_DEVICE_NONE;
+
+ sp<AudioOutputDescriptor> outputDesc = mOutputs.valueFor(output);
+
+ ssize_t index = mAudioPatches.indexOfKey(outputDesc->mPatchHandle);
+ if (index >= 0) {
+ sp<AudioPatch> patchDesc = mAudioPatches.valueAt(index);
+ if (patchDesc->mUid != mUidCached) {
+ ALOGV("getNewOutputDevice() device %08x forced by patch %d",
+ outputDesc->device(), outputDesc->mPatchHandle);
+ return outputDesc->device();
+ }
+ }
+
+ // check the following by order of priority to request a routing change if necessary:
+ // 1: the strategy enforced audible is active and enforced on the output:
+ // use device for strategy enforced audible
+ // 2: we are in call or the strategy phone is active on the output:
+ // use device for strategy phone
+ // 3: the strategy for enforced audible is active but not enforced on the output:
+ // use the device for strategy enforced audible
+ // 4: the strategy sonification is active on the output:
+ // use device for strategy sonification
+ // 5: the strategy "respectful" sonification is active on the output:
+ // use device for strategy "respectful" sonification
+ // 6: the strategy accessibility is active on the output:
+ // use device for strategy accessibility
+ // 7: the strategy media is active on the output:
+ // use device for strategy media
+ // 8: the strategy DTMF is active on the output:
+ // use device for strategy DTMF
+ // 9: the strategy for beacon, a.k.a. "transmitted through speaker" is active on the output:
+ // use device for strategy t-t-s
+ if (outputDesc->isStrategyActive(STRATEGY_ENFORCED_AUDIBLE) &&
+ mForceUse[AUDIO_POLICY_FORCE_FOR_SYSTEM] == AUDIO_POLICY_FORCE_SYSTEM_ENFORCED) {
+ device = getDeviceForStrategy(STRATEGY_ENFORCED_AUDIBLE, fromCache);
+ } else if (isInCall() ||
+ outputDesc->isStrategyActive(STRATEGY_PHONE)) {
+ device = getDeviceForStrategy(STRATEGY_PHONE, fromCache);
+ } else if (outputDesc->isStrategyActive(STRATEGY_ENFORCED_AUDIBLE)) {
+ device = getDeviceForStrategy(STRATEGY_ENFORCED_AUDIBLE, fromCache);
+ } else if (outputDesc->isStrategyActive(STRATEGY_SONIFICATION)) {
+ device = getDeviceForStrategy(STRATEGY_SONIFICATION, fromCache);
+ } else if (outputDesc->isStrategyActive(STRATEGY_SONIFICATION_RESPECTFUL)) {
+ device = getDeviceForStrategy(STRATEGY_SONIFICATION_RESPECTFUL, fromCache);
+ } else if (outputDesc->isStrategyActive(STRATEGY_ACCESSIBILITY)) {
+ device = getDeviceForStrategy(STRATEGY_ACCESSIBILITY, fromCache);
+ } else if (outputDesc->isStrategyActive(STRATEGY_MEDIA)) {
+ device = getDeviceForStrategy(STRATEGY_MEDIA, fromCache);
+ } else if (outputDesc->isStrategyActive(STRATEGY_DTMF)) {
+ device = getDeviceForStrategy(STRATEGY_DTMF, fromCache);
+ } else if (outputDesc->isStrategyActive(STRATEGY_TRANSMITTED_THROUGH_SPEAKER)) {
+ device = getDeviceForStrategy(STRATEGY_TRANSMITTED_THROUGH_SPEAKER, fromCache);
+ } else if (outputDesc->isStrategyActive(STRATEGY_REROUTING)) {
+ device = getDeviceForStrategy(STRATEGY_REROUTING, fromCache);
+ }
+
+ ALOGV("getNewOutputDevice() selected device %x", device);
+ return device;
+}
+
+audio_devices_t AudioPolicyManager::getNewInputDevice(audio_io_handle_t input)
+{
+ sp<AudioInputDescriptor> inputDesc = mInputs.valueFor(input);
+
+ ssize_t index = mAudioPatches.indexOfKey(inputDesc->mPatchHandle);
+ if (index >= 0) {
+ sp<AudioPatch> patchDesc = mAudioPatches.valueAt(index);
+ if (patchDesc->mUid != mUidCached) {
+ ALOGV("getNewInputDevice() device %08x forced by patch %d",
+ inputDesc->mDevice, inputDesc->mPatchHandle);
+ return inputDesc->mDevice;
+ }
+ }
+
+ audio_devices_t device = getDeviceAndMixForInputSource(inputDesc->mInputSource);
+
+ ALOGV("getNewInputDevice() selected device %x", device);
+ return device;
+}
+
+uint32_t AudioPolicyManager::getStrategyForStream(audio_stream_type_t stream) {
+ return (uint32_t)getStrategy(stream);
+}
+
+audio_devices_t AudioPolicyManager::getDevicesForStream(audio_stream_type_t stream) {
+ // By checking the range of stream before calling getStrategy, we avoid
+ // getStrategy's behavior for invalid streams. getStrategy would do a ALOGE
+ // and then return STRATEGY_MEDIA, but we want to return the empty set.
+ if (stream < (audio_stream_type_t) 0 || stream >= AUDIO_STREAM_PUBLIC_CNT) {
+ return AUDIO_DEVICE_NONE;
+ }
+ audio_devices_t devices;
+ routing_strategy strategy = getStrategy(stream);
+ devices = getDeviceForStrategy(strategy, true /*fromCache*/);
+ SortedVector<audio_io_handle_t> outputs = getOutputsForDevice(devices, mOutputs);
+ for (size_t i = 0; i < outputs.size(); i++) {
+ sp<AudioOutputDescriptor> outputDesc = mOutputs.valueFor(outputs[i]);
+ if (outputDesc->isStrategyActive(strategy)) {
+ devices = outputDesc->device();
+ break;
+ }
+ }
+
+ /*Filter SPEAKER_SAFE out of results, as AudioService doesn't know about it
+ and doesn't really need to.*/
+ if (devices & AUDIO_DEVICE_OUT_SPEAKER_SAFE) {
+ devices |= AUDIO_DEVICE_OUT_SPEAKER;
+ devices &= ~AUDIO_DEVICE_OUT_SPEAKER_SAFE;
+ }
+
+ return devices;
+}
+
+routing_strategy AudioPolicyManager::getStrategy(
+ audio_stream_type_t stream) {
+
+ ALOG_ASSERT(stream != AUDIO_STREAM_PATCH,"getStrategy() called for AUDIO_STREAM_PATCH");
+
+ // stream to strategy mapping
+ switch (stream) {
+ case AUDIO_STREAM_VOICE_CALL:
+ case AUDIO_STREAM_BLUETOOTH_SCO:
+ return STRATEGY_PHONE;
+ case AUDIO_STREAM_RING:
+ case AUDIO_STREAM_ALARM:
+ return STRATEGY_SONIFICATION;
+ case AUDIO_STREAM_NOTIFICATION:
+ return STRATEGY_SONIFICATION_RESPECTFUL;
+ case AUDIO_STREAM_DTMF:
+ return STRATEGY_DTMF;
+ default:
+ ALOGE("unknown stream type %d", stream);
+ case AUDIO_STREAM_SYSTEM:
+ // NOTE: SYSTEM stream uses MEDIA strategy because muting music and switching outputs
+ // while key clicks are played produces a poor result
+ case AUDIO_STREAM_MUSIC:
+ return STRATEGY_MEDIA;
+ case AUDIO_STREAM_ENFORCED_AUDIBLE:
+ return STRATEGY_ENFORCED_AUDIBLE;
+ case AUDIO_STREAM_TTS:
+ return STRATEGY_TRANSMITTED_THROUGH_SPEAKER;
+ case AUDIO_STREAM_ACCESSIBILITY:
+ return STRATEGY_ACCESSIBILITY;
+ case AUDIO_STREAM_REROUTING:
+ return STRATEGY_REROUTING;
+ }
+}
+
+uint32_t AudioPolicyManager::getStrategyForAttr(const audio_attributes_t *attr) {
+ // flags to strategy mapping
+ if ((attr->flags & AUDIO_FLAG_BEACON) == AUDIO_FLAG_BEACON) {
+ return (uint32_t) STRATEGY_TRANSMITTED_THROUGH_SPEAKER;
+ }
+ if ((attr->flags & AUDIO_FLAG_AUDIBILITY_ENFORCED) == AUDIO_FLAG_AUDIBILITY_ENFORCED) {
+ return (uint32_t) STRATEGY_ENFORCED_AUDIBLE;
+ }
+
+ // usage to strategy mapping
+ switch (attr->usage) {
+ case AUDIO_USAGE_ASSISTANCE_ACCESSIBILITY:
+ if (isStreamActive(AUDIO_STREAM_RING) || isStreamActive(AUDIO_STREAM_ALARM)) {
+ return (uint32_t) STRATEGY_SONIFICATION;
+ }
+ if (isInCall()) {
+ return (uint32_t) STRATEGY_PHONE;
+ }
+ return (uint32_t) STRATEGY_ACCESSIBILITY;
+
+ case AUDIO_USAGE_MEDIA:
+ case AUDIO_USAGE_GAME:
+ case AUDIO_USAGE_ASSISTANCE_NAVIGATION_GUIDANCE:
+ case AUDIO_USAGE_ASSISTANCE_SONIFICATION:
+ return (uint32_t) STRATEGY_MEDIA;
+
+ case AUDIO_USAGE_VOICE_COMMUNICATION:
+ return (uint32_t) STRATEGY_PHONE;
+
+ case AUDIO_USAGE_VOICE_COMMUNICATION_SIGNALLING:
+ return (uint32_t) STRATEGY_DTMF;
+
+ case AUDIO_USAGE_ALARM:
+ case AUDIO_USAGE_NOTIFICATION_TELEPHONY_RINGTONE:
+ return (uint32_t) STRATEGY_SONIFICATION;
+
+ case AUDIO_USAGE_NOTIFICATION:
+ case AUDIO_USAGE_NOTIFICATION_COMMUNICATION_REQUEST:
+ case AUDIO_USAGE_NOTIFICATION_COMMUNICATION_INSTANT:
+ case AUDIO_USAGE_NOTIFICATION_COMMUNICATION_DELAYED:
+ case AUDIO_USAGE_NOTIFICATION_EVENT:
+ return (uint32_t) STRATEGY_SONIFICATION_RESPECTFUL;
+
+ case AUDIO_USAGE_UNKNOWN:
+ default:
+ return (uint32_t) STRATEGY_MEDIA;
+ }
+}
+
+void AudioPolicyManager::handleNotificationRoutingForStream(audio_stream_type_t stream) {
+ switch(stream) {
+ case AUDIO_STREAM_MUSIC:
+ checkOutputForStrategy(STRATEGY_SONIFICATION_RESPECTFUL);
+ updateDevicesAndOutputs();
+ break;
+ default:
+ break;
+ }
+}
+
+bool AudioPolicyManager::isAnyOutputActive(audio_stream_type_t streamToIgnore) {
+ for (size_t s = 0 ; s < AUDIO_STREAM_CNT ; s++) {
+ if (s == (size_t) streamToIgnore) {
+ continue;
+ }
+ for (size_t i = 0; i < mOutputs.size(); i++) {
+ const sp<AudioOutputDescriptor> outputDesc = mOutputs.valueAt(i);
+ if (outputDesc->mRefCount[s] != 0) {
+ return true;
+ }
+ }
+ }
+ return false;
+}
+
+uint32_t AudioPolicyManager::handleEventForBeacon(int event) {
+ switch(event) {
+ case STARTING_OUTPUT:
+ mBeaconMuteRefCount++;
+ break;
+ case STOPPING_OUTPUT:
+ if (mBeaconMuteRefCount > 0) {
+ mBeaconMuteRefCount--;
+ }
+ break;
+ case STARTING_BEACON:
+ mBeaconPlayingRefCount++;
+ break;
+ case STOPPING_BEACON:
+ if (mBeaconPlayingRefCount > 0) {
+ mBeaconPlayingRefCount--;
+ }
+ break;
+ }
+
+ if (mBeaconMuteRefCount > 0) {
+ // any playback causes beacon to be muted
+ return setBeaconMute(true);
+ } else {
+ // no other playback: unmute when beacon starts playing, mute when it stops
+ return setBeaconMute(mBeaconPlayingRefCount == 0);
+ }
+}
+
+uint32_t AudioPolicyManager::setBeaconMute(bool mute) {
+ ALOGV("setBeaconMute(%d) mBeaconMuteRefCount=%d mBeaconPlayingRefCount=%d",
+ mute, mBeaconMuteRefCount, mBeaconPlayingRefCount);
+ // keep track of muted state to avoid repeating mute/unmute operations
+ if (mBeaconMuted != mute) {
+ // mute/unmute AUDIO_STREAM_TTS on all outputs
+ ALOGV("\t muting %d", mute);
+ uint32_t maxLatency = 0;
+ for (size_t i = 0; i < mOutputs.size(); i++) {
+ sp<AudioOutputDescriptor> desc = mOutputs.valueAt(i);
+ setStreamMute(AUDIO_STREAM_TTS, mute/*on*/,
+ desc->mIoHandle,
+ 0 /*delay*/, AUDIO_DEVICE_NONE);
+ const uint32_t latency = desc->latency() * 2;
+ if (latency > maxLatency) {
+ maxLatency = latency;
+ }
+ }
+ mBeaconMuted = mute;
+ return maxLatency;
+ }
+ return 0;
+}
+
+audio_devices_t AudioPolicyManager::getDeviceForStrategy(routing_strategy strategy,
+ bool fromCache)
+{
+ uint32_t device = AUDIO_DEVICE_NONE;
+
+ if (fromCache) {
+ ALOGVV("getDeviceForStrategy() from cache strategy %d, device %x",
+ strategy, mDeviceForStrategy[strategy]);
+ return mDeviceForStrategy[strategy];
+ }
+ audio_devices_t availableOutputDeviceTypes = mAvailableOutputDevices.types();
+ switch (strategy) {
+
+ case STRATEGY_TRANSMITTED_THROUGH_SPEAKER:
+ device = availableOutputDeviceTypes & AUDIO_DEVICE_OUT_SPEAKER;
+ if (!device) {
+ ALOGE("getDeviceForStrategy() no device found for "\
+ "STRATEGY_TRANSMITTED_THROUGH_SPEAKER");
+ }
+ break;
+
+ case STRATEGY_SONIFICATION_RESPECTFUL:
+ if (isInCall()) {
+ device = getDeviceForStrategy(STRATEGY_SONIFICATION, false /*fromCache*/);
+ } else if (isStreamActiveRemotely(AUDIO_STREAM_MUSIC,
+ SONIFICATION_RESPECTFUL_AFTER_MUSIC_DELAY)) {
+ // while media is playing on a remote device, use the the sonification behavior.
+ // Note that we test this usecase before testing if media is playing because
+ // the isStreamActive() method only informs about the activity of a stream, not
+ // if it's for local playback. Note also that we use the same delay between both tests
+ device = getDeviceForStrategy(STRATEGY_SONIFICATION, false /*fromCache*/);
+ //user "safe" speaker if available instead of normal speaker to avoid triggering
+ //other acoustic safety mechanisms for notification
+ if (device == AUDIO_DEVICE_OUT_SPEAKER && (availableOutputDeviceTypes & AUDIO_DEVICE_OUT_SPEAKER_SAFE))
+ device = AUDIO_DEVICE_OUT_SPEAKER_SAFE;
+ } else if (isStreamActive(AUDIO_STREAM_MUSIC, SONIFICATION_RESPECTFUL_AFTER_MUSIC_DELAY)) {
+ // while media is playing (or has recently played), use the same device
+ device = getDeviceForStrategy(STRATEGY_MEDIA, false /*fromCache*/);
+ } else {
+ // when media is not playing anymore, fall back on the sonification behavior
+ device = getDeviceForStrategy(STRATEGY_SONIFICATION, false /*fromCache*/);
+ //user "safe" speaker if available instead of normal speaker to avoid triggering
+ //other acoustic safety mechanisms for notification
+ if (device == AUDIO_DEVICE_OUT_SPEAKER && (availableOutputDeviceTypes & AUDIO_DEVICE_OUT_SPEAKER_SAFE))
+ device = AUDIO_DEVICE_OUT_SPEAKER_SAFE;
+ }
+
+ break;
+
+ case STRATEGY_DTMF:
+ if (!isInCall()) {
+ // when off call, DTMF strategy follows the same rules as MEDIA strategy
+ device = getDeviceForStrategy(STRATEGY_MEDIA, false /*fromCache*/);
+ break;
+ }
+ // when in call, DTMF and PHONE strategies follow the same rules
+ // FALL THROUGH
+
+ case STRATEGY_PHONE:
+ // Force use of only devices on primary output if:
+ // - in call AND
+ // - cannot route from voice call RX OR
+ // - audio HAL version is < 3.0 and TX device is on the primary HW module
+ if (mPhoneState == AUDIO_MODE_IN_CALL) {
+ audio_devices_t txDevice =
+ getDeviceAndMixForInputSource(AUDIO_SOURCE_VOICE_COMMUNICATION);
+ sp<AudioOutputDescriptor> hwOutputDesc = mOutputs.valueFor(mPrimaryOutput);
+ if (((mAvailableInputDevices.types() &
+ AUDIO_DEVICE_IN_TELEPHONY_RX & ~AUDIO_DEVICE_BIT_IN) == 0) ||
+ (((txDevice & availablePrimaryInputDevices() & ~AUDIO_DEVICE_BIT_IN) != 0) &&
+ (hwOutputDesc->getAudioPort()->mModule->mHalVersion <
+ AUDIO_DEVICE_API_VERSION_3_0))) {
+ availableOutputDeviceTypes = availablePrimaryOutputDevices();
+ }
+ }
+ // for phone strategy, we first consider the forced use and then the available devices by order
+ // of priority
+ switch (mForceUse[AUDIO_POLICY_FORCE_FOR_COMMUNICATION]) {
+ case AUDIO_POLICY_FORCE_BT_SCO:
+ if (!isInCall() || strategy != STRATEGY_DTMF) {
+ device = availableOutputDeviceTypes & AUDIO_DEVICE_OUT_BLUETOOTH_SCO_CARKIT;
+ if (device) break;
+ }
+ device = availableOutputDeviceTypes & AUDIO_DEVICE_OUT_BLUETOOTH_SCO_HEADSET;
+ if (device) break;
+ device = availableOutputDeviceTypes & AUDIO_DEVICE_OUT_BLUETOOTH_SCO;
+ if (device) break;
+ // if SCO device is requested but no SCO device is available, fall back to default case
+ // FALL THROUGH
+
+ default: // FORCE_NONE
+ // when not in a phone call, phone strategy should route STREAM_VOICE_CALL to A2DP
+ if (!isInCall() &&
+ (mForceUse[AUDIO_POLICY_FORCE_FOR_MEDIA] != AUDIO_POLICY_FORCE_NO_BT_A2DP) &&
+ (getA2dpOutput() != 0)) {
+ device = availableOutputDeviceTypes & AUDIO_DEVICE_OUT_BLUETOOTH_A2DP;
+ if (device) break;
+ device = availableOutputDeviceTypes & AUDIO_DEVICE_OUT_BLUETOOTH_A2DP_HEADPHONES;
+ if (device) break;
+ }
+ device = availableOutputDeviceTypes & AUDIO_DEVICE_OUT_WIRED_HEADPHONE;
+ if (device) break;
+ device = availableOutputDeviceTypes & AUDIO_DEVICE_OUT_WIRED_HEADSET;
+ if (device) break;
+ device = availableOutputDeviceTypes & AUDIO_DEVICE_OUT_USB_DEVICE;
+ if (device) break;
+ if (mPhoneState != AUDIO_MODE_IN_CALL) {
+ device = availableOutputDeviceTypes & AUDIO_DEVICE_OUT_USB_ACCESSORY;
+ if (device) break;
+ device = availableOutputDeviceTypes & AUDIO_DEVICE_OUT_DGTL_DOCK_HEADSET;
+ if (device) break;
+ device = availableOutputDeviceTypes & AUDIO_DEVICE_OUT_AUX_DIGITAL;
+ if (device) break;
+ device = availableOutputDeviceTypes & AUDIO_DEVICE_OUT_ANLG_DOCK_HEADSET;
+ if (device) break;
+ }
+ device = availableOutputDeviceTypes & AUDIO_DEVICE_OUT_EARPIECE;
+ if (device) break;
+ device = mDefaultOutputDevice->mDeviceType;
+ if (device == AUDIO_DEVICE_NONE) {
+ ALOGE("getDeviceForStrategy() no device found for STRATEGY_PHONE");
+ }
+ break;
+
+ case AUDIO_POLICY_FORCE_SPEAKER:
+ // when not in a phone call, phone strategy should route STREAM_VOICE_CALL to
+ // A2DP speaker when forcing to speaker output
+ if (!isInCall() &&
+ (mForceUse[AUDIO_POLICY_FORCE_FOR_MEDIA] != AUDIO_POLICY_FORCE_NO_BT_A2DP) &&
+ (getA2dpOutput() != 0)) {
+ device = availableOutputDeviceTypes & AUDIO_DEVICE_OUT_BLUETOOTH_A2DP_SPEAKER;
+ if (device) break;
+ }
+ if (!isInCall()) {
+ device = availableOutputDeviceTypes & AUDIO_DEVICE_OUT_USB_ACCESSORY;
+ if (device) break;
+ device = availableOutputDeviceTypes & AUDIO_DEVICE_OUT_USB_DEVICE;
+ if (device) break;
+ device = availableOutputDeviceTypes & AUDIO_DEVICE_OUT_DGTL_DOCK_HEADSET;
+ if (device) break;
+ device = availableOutputDeviceTypes & AUDIO_DEVICE_OUT_AUX_DIGITAL;
+ if (device) break;
+ device = availableOutputDeviceTypes & AUDIO_DEVICE_OUT_ANLG_DOCK_HEADSET;
+ if (device) break;
+ }
+ device = availableOutputDeviceTypes & AUDIO_DEVICE_OUT_LINE;
+ if (device) break;
+ device = availableOutputDeviceTypes & AUDIO_DEVICE_OUT_SPEAKER;
+ if (device) break;
+ device = mDefaultOutputDevice->mDeviceType;
+ if (device == AUDIO_DEVICE_NONE) {
+ ALOGE("getDeviceForStrategy() no device found for STRATEGY_PHONE, FORCE_SPEAKER");
+ }
+ break;
+ }
+ break;
+
+ case STRATEGY_SONIFICATION:
+
+ // If incall, just select the STRATEGY_PHONE device: The rest of the behavior is handled by
+ // handleIncallSonification().
+ if (isInCall()) {
+ device = getDeviceForStrategy(STRATEGY_PHONE, false /*fromCache*/);
+ break;
+ }
+ // FALL THROUGH
+
+ case STRATEGY_ENFORCED_AUDIBLE:
+ // strategy STRATEGY_ENFORCED_AUDIBLE uses same routing policy as STRATEGY_SONIFICATION
+ // except:
+ // - when in call where it doesn't default to STRATEGY_PHONE behavior
+ // - in countries where not enforced in which case it follows STRATEGY_MEDIA
+
+ if ((strategy == STRATEGY_SONIFICATION) ||
+ (mForceUse[AUDIO_POLICY_FORCE_FOR_SYSTEM] == AUDIO_POLICY_FORCE_SYSTEM_ENFORCED)) {
+ device = availableOutputDeviceTypes & AUDIO_DEVICE_OUT_SPEAKER;
+ if (device == AUDIO_DEVICE_NONE) {
+ ALOGE("getDeviceForStrategy() speaker device not found for STRATEGY_SONIFICATION");
+ }
+ }
+ // The second device used for sonification is the same as the device used by media strategy
+ // FALL THROUGH
+
+ // FIXME: STRATEGY_ACCESSIBILITY and STRATEGY_REROUTING follow STRATEGY_MEDIA for now
+ case STRATEGY_ACCESSIBILITY:
+ if (strategy == STRATEGY_ACCESSIBILITY) {
+ // do not route accessibility prompts to a digital output currently configured with a
+ // compressed format as they would likely not be mixed and dropped.
+ for (size_t i = 0; i < mOutputs.size(); i++) {
+ sp<AudioOutputDescriptor> desc = mOutputs.valueAt(i);
+ audio_devices_t devices = desc->device() &
+ (AUDIO_DEVICE_OUT_HDMI | AUDIO_DEVICE_OUT_SPDIF | AUDIO_DEVICE_OUT_HDMI_ARC);
+ if (desc->isActive() && !audio_is_linear_pcm(desc->mFormat) &&
+ devices != AUDIO_DEVICE_NONE) {
+ availableOutputDeviceTypes = availableOutputDeviceTypes & ~devices;
+ }
+ }
+ }
+ // FALL THROUGH
+
+ case STRATEGY_REROUTING:
+ case STRATEGY_MEDIA: {
+ uint32_t device2 = AUDIO_DEVICE_NONE;
+ if (strategy != STRATEGY_SONIFICATION) {
+ // no sonification on remote submix (e.g. WFD)
+ if (mAvailableOutputDevices.getDevice(AUDIO_DEVICE_OUT_REMOTE_SUBMIX, String8("0")) != 0) {
+ device2 = availableOutputDeviceTypes & AUDIO_DEVICE_OUT_REMOTE_SUBMIX;
+ }
+ }
+ if ((device2 == AUDIO_DEVICE_NONE) &&
+ (mForceUse[AUDIO_POLICY_FORCE_FOR_MEDIA] != AUDIO_POLICY_FORCE_NO_BT_A2DP) &&
+ (getA2dpOutput() != 0)) {
+ device2 = availableOutputDeviceTypes & AUDIO_DEVICE_OUT_BLUETOOTH_A2DP;
+ if (device2 == AUDIO_DEVICE_NONE) {
+ device2 = availableOutputDeviceTypes & AUDIO_DEVICE_OUT_BLUETOOTH_A2DP_HEADPHONES;
+ }
+ if (device2 == AUDIO_DEVICE_NONE) {
+ device2 = availableOutputDeviceTypes & AUDIO_DEVICE_OUT_BLUETOOTH_A2DP_SPEAKER;
+ }
+ }
+ if ((device2 == AUDIO_DEVICE_NONE) &&
+ (mForceUse[AUDIO_POLICY_FORCE_FOR_MEDIA] == AUDIO_POLICY_FORCE_SPEAKER)) {
+ device2 = availableOutputDeviceTypes & AUDIO_DEVICE_OUT_SPEAKER;
+ }
+ if (device2 == AUDIO_DEVICE_NONE) {
+ device2 = availableOutputDeviceTypes & AUDIO_DEVICE_OUT_WIRED_HEADPHONE;
+ }
+ if ((device2 == AUDIO_DEVICE_NONE)) {
+ device2 = availableOutputDeviceTypes & AUDIO_DEVICE_OUT_LINE;
+ }
+ if (device2 == AUDIO_DEVICE_NONE) {
+ device2 = availableOutputDeviceTypes & AUDIO_DEVICE_OUT_WIRED_HEADSET;
+ }
+ if (device2 == AUDIO_DEVICE_NONE) {
+ device2 = availableOutputDeviceTypes & AUDIO_DEVICE_OUT_USB_ACCESSORY;
+ }
+ if (device2 == AUDIO_DEVICE_NONE) {
+ device2 = availableOutputDeviceTypes & AUDIO_DEVICE_OUT_USB_DEVICE;
+ }
+ if (device2 == AUDIO_DEVICE_NONE) {
+ device2 = availableOutputDeviceTypes & AUDIO_DEVICE_OUT_DGTL_DOCK_HEADSET;
+ }
+ if ((device2 == AUDIO_DEVICE_NONE) && (strategy != STRATEGY_SONIFICATION)) {
+ // no sonification on aux digital (e.g. HDMI)
+ device2 = availableOutputDeviceTypes & AUDIO_DEVICE_OUT_AUX_DIGITAL;
+ }
+ if ((device2 == AUDIO_DEVICE_NONE) &&
+ (mForceUse[AUDIO_POLICY_FORCE_FOR_DOCK] == AUDIO_POLICY_FORCE_ANALOG_DOCK)) {
+ device2 = availableOutputDeviceTypes & AUDIO_DEVICE_OUT_ANLG_DOCK_HEADSET;
+ }
+ if (device2 == AUDIO_DEVICE_NONE) {
+ device2 = availableOutputDeviceTypes & AUDIO_DEVICE_OUT_SPEAKER;
+ }
+ int device3 = AUDIO_DEVICE_NONE;
+ if (strategy == STRATEGY_MEDIA) {
+ // ARC, SPDIF and AUX_LINE can co-exist with others.
+ device3 = availableOutputDeviceTypes & AUDIO_DEVICE_OUT_HDMI_ARC;
+ device3 |= (availableOutputDeviceTypes & AUDIO_DEVICE_OUT_SPDIF);
+ device3 |= (availableOutputDeviceTypes & AUDIO_DEVICE_OUT_AUX_LINE);
+ }
+
+ device2 |= device3;
+ // device is DEVICE_OUT_SPEAKER if we come from case STRATEGY_SONIFICATION or
+ // STRATEGY_ENFORCED_AUDIBLE, AUDIO_DEVICE_NONE otherwise
+ device |= device2;
+
+ // If hdmi system audio mode is on, remove speaker out of output list.
+ if ((strategy == STRATEGY_MEDIA) &&
+ (mForceUse[AUDIO_POLICY_FORCE_FOR_HDMI_SYSTEM_AUDIO] ==
+ AUDIO_POLICY_FORCE_HDMI_SYSTEM_AUDIO_ENFORCED)) {
+ device &= ~AUDIO_DEVICE_OUT_SPEAKER;
+ }
+
+ if (device) break;
+ device = mDefaultOutputDevice->mDeviceType;
+ if (device == AUDIO_DEVICE_NONE) {
+ ALOGE("getDeviceForStrategy() no device found for STRATEGY_MEDIA");
+ }
+ } break;
+
+ default:
+ ALOGW("getDeviceForStrategy() unknown strategy: %d", strategy);
+ break;
+ }
+
+ ALOGVV("getDeviceForStrategy() strategy %d, device %x", strategy, device);
+ return device;
+}
+
+void AudioPolicyManager::updateDevicesAndOutputs()
+{
+ for (int i = 0; i < NUM_STRATEGIES; i++) {
+ mDeviceForStrategy[i] = getDeviceForStrategy((routing_strategy)i, false /*fromCache*/);
+ }
+ mPreviousOutputs = mOutputs;
+}
+
+uint32_t AudioPolicyManager::checkDeviceMuteStrategies(sp<AudioOutputDescriptor> outputDesc,
+ audio_devices_t prevDevice,
+ uint32_t delayMs)
+{
+ // mute/unmute strategies using an incompatible device combination
+ // if muting, wait for the audio in pcm buffer to be drained before proceeding
+ // if unmuting, unmute only after the specified delay
+ if (outputDesc->isDuplicated()) {
+ return 0;
+ }
+
+ uint32_t muteWaitMs = 0;
+ audio_devices_t device = outputDesc->device();
+ bool shouldMute = outputDesc->isActive() && (popcount(device) >= 2);
+
+ for (size_t i = 0; i < NUM_STRATEGIES; i++) {
+ audio_devices_t curDevice = getDeviceForStrategy((routing_strategy)i, false /*fromCache*/);
+ curDevice = curDevice & outputDesc->mProfile->mSupportedDevices.types();
+ bool mute = shouldMute && (curDevice & device) && (curDevice != device);
+ bool doMute = false;
+
+ if (mute && !outputDesc->mStrategyMutedByDevice[i]) {
+ doMute = true;
+ outputDesc->mStrategyMutedByDevice[i] = true;
+ } else if (!mute && outputDesc->mStrategyMutedByDevice[i]){
+ doMute = true;
+ outputDesc->mStrategyMutedByDevice[i] = false;
+ }
+ if (doMute) {
+ for (size_t j = 0; j < mOutputs.size(); j++) {
+ sp<AudioOutputDescriptor> desc = mOutputs.valueAt(j);
+ // skip output if it does not share any device with current output
+ if ((desc->supportedDevices() & outputDesc->supportedDevices())
+ == AUDIO_DEVICE_NONE) {
+ continue;
+ }
+ audio_io_handle_t curOutput = mOutputs.keyAt(j);
+ ALOGVV("checkDeviceMuteStrategies() %s strategy %d (curDevice %04x) on output %d",
+ mute ? "muting" : "unmuting", i, curDevice, curOutput);
+ setStrategyMute((routing_strategy)i, mute, curOutput, mute ? 0 : delayMs);
+ if (desc->isStrategyActive((routing_strategy)i)) {
+ if (mute) {
+ // FIXME: should not need to double latency if volume could be applied
+ // immediately by the audioflinger mixer. We must account for the delay
+ // between now and the next time the audioflinger thread for this output
+ // will process a buffer (which corresponds to one buffer size,
+ // usually 1/2 or 1/4 of the latency).
+ if (muteWaitMs < desc->latency() * 2) {
+ muteWaitMs = desc->latency() * 2;
+ }
+ }
+ }
+ }
+ }
+ }
+
+ // temporary mute output if device selection changes to avoid volume bursts due to
+ // different per device volumes
+ if (outputDesc->isActive() && (device != prevDevice)) {
+ if (muteWaitMs < outputDesc->latency() * 2) {
+ muteWaitMs = outputDesc->latency() * 2;
+ }
+ for (size_t i = 0; i < NUM_STRATEGIES; i++) {
+ if (outputDesc->isStrategyActive((routing_strategy)i)) {
+ setStrategyMute((routing_strategy)i, true, outputDesc->mIoHandle);
+ // do tempMute unmute after twice the mute wait time
+ setStrategyMute((routing_strategy)i, false, outputDesc->mIoHandle,
+ muteWaitMs *2, device);
+ }
+ }
+ }
+
+ // wait for the PCM output buffers to empty before proceeding with the rest of the command
+ if (muteWaitMs > delayMs) {
+ muteWaitMs -= delayMs;
+ usleep(muteWaitMs * 1000);
+ return muteWaitMs;
+ }
+ return 0;
+}
+
+uint32_t AudioPolicyManager::setOutputDevice(audio_io_handle_t output,
+ audio_devices_t device,
+ bool force,
+ int delayMs,
+ audio_patch_handle_t *patchHandle,
+ const char* address)
+{
+ ALOGV("setOutputDevice() output %d device %04x delayMs %d", output, device, delayMs);
+ sp<AudioOutputDescriptor> outputDesc = mOutputs.valueFor(output);
+ AudioParameter param;
+ uint32_t muteWaitMs;
+
+ if (outputDesc->isDuplicated()) {
+ muteWaitMs = setOutputDevice(outputDesc->mOutput1->mIoHandle, device, force, delayMs);
+ muteWaitMs += setOutputDevice(outputDesc->mOutput2->mIoHandle, device, force, delayMs);
+ return muteWaitMs;
+ }
+ // no need to proceed if new device is not AUDIO_DEVICE_NONE and not supported by current
+ // output profile
+ if ((device != AUDIO_DEVICE_NONE) &&
+ ((device & outputDesc->mProfile->mSupportedDevices.types()) == 0)) {
+ return 0;
+ }
+
+ // filter devices according to output selected
+ device = (audio_devices_t)(device & outputDesc->mProfile->mSupportedDevices.types());
+
+ audio_devices_t prevDevice = outputDesc->mDevice;
+
+ ALOGV("setOutputDevice() prevDevice %04x", prevDevice);
+
+ if (device != AUDIO_DEVICE_NONE) {
+ outputDesc->mDevice = device;
+ }
+ muteWaitMs = checkDeviceMuteStrategies(outputDesc, prevDevice, delayMs);
+
+ // Do not change the routing if:
+ // the requested device is AUDIO_DEVICE_NONE
+ // OR the requested device is the same as current device
+ // AND force is not specified
+ // AND the output is connected by a valid audio patch.
+ // Doing this check here allows the caller to call setOutputDevice() without conditions
+ if ((device == AUDIO_DEVICE_NONE || device == prevDevice) && !force &&
+ outputDesc->mPatchHandle != 0) {
+ ALOGV("setOutputDevice() setting same device %04x or null device for output %d",
+ device, output);
+ return muteWaitMs;
+ }
+
+ ALOGV("setOutputDevice() changing device");
+
+ // do the routing
+ if (device == AUDIO_DEVICE_NONE) {
+ resetOutputDevice(output, delayMs, NULL);
+ } else {
+ DeviceVector deviceList = (address == NULL) ?
+ mAvailableOutputDevices.getDevicesFromType(device)
+ : mAvailableOutputDevices.getDevicesFromTypeAddr(device, String8(address));
+ if (!deviceList.isEmpty()) {
+ struct audio_patch patch;
+ outputDesc->toAudioPortConfig(&patch.sources[0]);
+ patch.num_sources = 1;
+ patch.num_sinks = 0;
+ for (size_t i = 0; i < deviceList.size() && i < AUDIO_PATCH_PORTS_MAX; i++) {
+ deviceList.itemAt(i)->toAudioPortConfig(&patch.sinks[i]);
+ patch.num_sinks++;
+ }
+ ssize_t index;
+ if (patchHandle && *patchHandle != AUDIO_PATCH_HANDLE_NONE) {
+ index = mAudioPatches.indexOfKey(*patchHandle);
+ } else {
+ index = mAudioPatches.indexOfKey(outputDesc->mPatchHandle);
+ }
+ sp< AudioPatch> patchDesc;
+ audio_patch_handle_t afPatchHandle = AUDIO_PATCH_HANDLE_NONE;
+ if (index >= 0) {
+ patchDesc = mAudioPatches.valueAt(index);
+ afPatchHandle = patchDesc->mAfPatchHandle;
+ }
+
+ status_t status = mpClientInterface->createAudioPatch(&patch,
+ &afPatchHandle,
+ delayMs);
+ ALOGV("setOutputDevice() createAudioPatch returned %d patchHandle %d"
+ "num_sources %d num_sinks %d",
+ status, afPatchHandle, patch.num_sources, patch.num_sinks);
+ if (status == NO_ERROR) {
+ if (index < 0) {
+ patchDesc = new AudioPatch((audio_patch_handle_t)nextUniqueId(),
+ &patch, mUidCached);
+ addAudioPatch(patchDesc->mHandle, patchDesc);
+ } else {
+ patchDesc->mPatch = patch;
+ }
+ patchDesc->mAfPatchHandle = afPatchHandle;
+ patchDesc->mUid = mUidCached;
+ if (patchHandle) {
+ *patchHandle = patchDesc->mHandle;
+ }
+ outputDesc->mPatchHandle = patchDesc->mHandle;
+ nextAudioPortGeneration();
+ mpClientInterface->onAudioPatchListUpdate();
+ }
+ }
+
+ // inform all input as well
+ for (size_t i = 0; i < mInputs.size(); i++) {
+ const sp<AudioInputDescriptor> inputDescriptor = mInputs.valueAt(i);
+ if (!isVirtualInputDevice(inputDescriptor->mDevice)) {
+ AudioParameter inputCmd = AudioParameter();
+ ALOGV("%s: inform input %d of device:%d", __func__,
+ inputDescriptor->mIoHandle, device);
+ inputCmd.addInt(String8(AudioParameter::keyRouting),device);
+ mpClientInterface->setParameters(inputDescriptor->mIoHandle,
+ inputCmd.toString(),
+ delayMs);
+ }
+ }
+ }
+
+ // update stream volumes according to new device
+ applyStreamVolumes(output, device, delayMs);
+
+ return muteWaitMs;
+}
+
+status_t AudioPolicyManager::resetOutputDevice(audio_io_handle_t output,
+ int delayMs,
+ audio_patch_handle_t *patchHandle)
+{
+ sp<AudioOutputDescriptor> outputDesc = mOutputs.valueFor(output);
+ ssize_t index;
+ if (patchHandle) {
+ index = mAudioPatches.indexOfKey(*patchHandle);
+ } else {
+ index = mAudioPatches.indexOfKey(outputDesc->mPatchHandle);
+ }
+ if (index < 0) {
+ return INVALID_OPERATION;
+ }
+ sp< AudioPatch> patchDesc = mAudioPatches.valueAt(index);
+ status_t status = mpClientInterface->releaseAudioPatch(patchDesc->mAfPatchHandle, delayMs);
+ ALOGV("resetOutputDevice() releaseAudioPatch returned %d", status);
+ outputDesc->mPatchHandle = 0;
+ removeAudioPatch(patchDesc->mHandle);
+ nextAudioPortGeneration();
+ mpClientInterface->onAudioPatchListUpdate();
+ return status;
+}
+
+status_t AudioPolicyManager::setInputDevice(audio_io_handle_t input,
+ audio_devices_t device,
+ bool force,
+ audio_patch_handle_t *patchHandle)
+{
+ status_t status = NO_ERROR;
+
+ sp<AudioInputDescriptor> inputDesc = mInputs.valueFor(input);
+ if ((device != AUDIO_DEVICE_NONE) && ((device != inputDesc->mDevice) || force)) {
+ inputDesc->mDevice = device;
+
+ DeviceVector deviceList = mAvailableInputDevices.getDevicesFromType(device);
+ if (!deviceList.isEmpty()) {
+ struct audio_patch patch;
+ inputDesc->toAudioPortConfig(&patch.sinks[0]);
+ // AUDIO_SOURCE_HOTWORD is for internal use only:
+ // handled as AUDIO_SOURCE_VOICE_RECOGNITION by the audio HAL
+ if (patch.sinks[0].ext.mix.usecase.source == AUDIO_SOURCE_HOTWORD &&
+ !inputDesc->mIsSoundTrigger) {
+ patch.sinks[0].ext.mix.usecase.source = AUDIO_SOURCE_VOICE_RECOGNITION;
+ }
+ patch.num_sinks = 1;
+ //only one input device for now
+ deviceList.itemAt(0)->toAudioPortConfig(&patch.sources[0]);
+ patch.num_sources = 1;
+ ssize_t index;
+ if (patchHandle && *patchHandle != AUDIO_PATCH_HANDLE_NONE) {
+ index = mAudioPatches.indexOfKey(*patchHandle);
+ } else {
+ index = mAudioPatches.indexOfKey(inputDesc->mPatchHandle);
+ }
+ sp< AudioPatch> patchDesc;
+ audio_patch_handle_t afPatchHandle = AUDIO_PATCH_HANDLE_NONE;
+ if (index >= 0) {
+ patchDesc = mAudioPatches.valueAt(index);
+ afPatchHandle = patchDesc->mAfPatchHandle;
+ }
+
+ status_t status = mpClientInterface->createAudioPatch(&patch,
+ &afPatchHandle,
+ 0);
+ ALOGV("setInputDevice() createAudioPatch returned %d patchHandle %d",
+ status, afPatchHandle);
+ if (status == NO_ERROR) {
+ if (index < 0) {
+ patchDesc = new AudioPatch((audio_patch_handle_t)nextUniqueId(),
+ &patch, mUidCached);
+ addAudioPatch(patchDesc->mHandle, patchDesc);
+ } else {
+ patchDesc->mPatch = patch;
+ }
+ patchDesc->mAfPatchHandle = afPatchHandle;
+ patchDesc->mUid = mUidCached;
+ if (patchHandle) {
+ *patchHandle = patchDesc->mHandle;
+ }
+ inputDesc->mPatchHandle = patchDesc->mHandle;
+ nextAudioPortGeneration();
+ mpClientInterface->onAudioPatchListUpdate();
+ }
+ }
+ }
+ return status;
+}
+
+status_t AudioPolicyManager::resetInputDevice(audio_io_handle_t input,
+ audio_patch_handle_t *patchHandle)
+{
+ sp<AudioInputDescriptor> inputDesc = mInputs.valueFor(input);
+ ssize_t index;
+ if (patchHandle) {
+ index = mAudioPatches.indexOfKey(*patchHandle);
+ } else {
+ index = mAudioPatches.indexOfKey(inputDesc->mPatchHandle);
+ }
+ if (index < 0) {
+ return INVALID_OPERATION;
+ }
+ sp< AudioPatch> patchDesc = mAudioPatches.valueAt(index);
+ status_t status = mpClientInterface->releaseAudioPatch(patchDesc->mAfPatchHandle, 0);
+ ALOGV("resetInputDevice() releaseAudioPatch returned %d", status);
+ inputDesc->mPatchHandle = 0;
+ removeAudioPatch(patchDesc->mHandle);
+ nextAudioPortGeneration();
+ mpClientInterface->onAudioPatchListUpdate();
+ return status;
+}
+
+sp<IOProfile> AudioPolicyManager::getInputProfile(audio_devices_t device,
+ String8 address,
+ uint32_t& samplingRate,
+ audio_format_t format,
+ audio_channel_mask_t channelMask,
+ audio_input_flags_t flags)
+{
+ // Choose an input profile based on the requested capture parameters: select the first available
+ // profile supporting all requested parameters.
+
+ for (size_t i = 0; i < mHwModules.size(); i++)
+ {
+ if (mHwModules[i]->mHandle == 0) {
+ continue;
+ }
+ for (size_t j = 0; j < mHwModules[i]->mInputProfiles.size(); j++)
+ {
+ sp<IOProfile> profile = mHwModules[i]->mInputProfiles[j];
+ // profile->log();
+ if (profile->isCompatibleProfile(device, address, samplingRate,
+ &samplingRate /*updatedSamplingRate*/,
+ format, channelMask, (audio_output_flags_t) flags)) {
+
+ return profile;
+ }
+ }
+ }
+ return NULL;
+}
+
+
+audio_devices_t AudioPolicyManager::getDeviceAndMixForInputSource(audio_source_t inputSource,
+ AudioMix **policyMix)
+{
+ audio_devices_t availableDeviceTypes = mAvailableInputDevices.types() &
+ ~AUDIO_DEVICE_BIT_IN;
+
+ for (size_t i = 0; i < mPolicyMixes.size(); i++) {
+ if (mPolicyMixes[i]->mMix.mMixType != MIX_TYPE_RECORDERS) {
+ continue;
+ }
+ for (size_t j = 0; j < mPolicyMixes[i]->mMix.mCriteria.size(); j++) {
+ if ((RULE_MATCH_ATTRIBUTE_CAPTURE_PRESET == mPolicyMixes[i]->mMix.mCriteria[j].mRule &&
+ mPolicyMixes[i]->mMix.mCriteria[j].mAttr.mSource == inputSource) ||
+ (RULE_EXCLUDE_ATTRIBUTE_CAPTURE_PRESET == mPolicyMixes[i]->mMix.mCriteria[j].mRule &&
+ mPolicyMixes[i]->mMix.mCriteria[j].mAttr.mSource != inputSource)) {
+ if (availableDeviceTypes & AUDIO_DEVICE_IN_REMOTE_SUBMIX) {
+ if (policyMix != NULL) {
+ *policyMix = &mPolicyMixes[i]->mMix;
+ }
+ return AUDIO_DEVICE_IN_REMOTE_SUBMIX;
+ }
+ break;
+ }
+ }
+ }
+
+ return getDeviceForInputSource(inputSource);
+}
+
+audio_devices_t AudioPolicyManager::getDeviceForInputSource(audio_source_t inputSource)
+{
+ uint32_t device = AUDIO_DEVICE_NONE;
+ audio_devices_t availableDeviceTypes = mAvailableInputDevices.types() &
+ ~AUDIO_DEVICE_BIT_IN;
+
+ switch (inputSource) {
+ case AUDIO_SOURCE_VOICE_UPLINK:
+ if (availableDeviceTypes & AUDIO_DEVICE_IN_VOICE_CALL) {
+ device = AUDIO_DEVICE_IN_VOICE_CALL;
+ break;
+ }
+ break;
+
+ case AUDIO_SOURCE_DEFAULT:
+ case AUDIO_SOURCE_MIC:
+ if (availableDeviceTypes & AUDIO_DEVICE_IN_BLUETOOTH_A2DP) {
+ device = AUDIO_DEVICE_IN_BLUETOOTH_A2DP;
+ } else if ((mForceUse[AUDIO_POLICY_FORCE_FOR_RECORD] == AUDIO_POLICY_FORCE_BT_SCO) &&
+ (availableDeviceTypes & AUDIO_DEVICE_IN_BLUETOOTH_SCO_HEADSET)) {
+ device = AUDIO_DEVICE_IN_BLUETOOTH_SCO_HEADSET;
+ } else if (availableDeviceTypes & AUDIO_DEVICE_IN_WIRED_HEADSET) {
+ device = AUDIO_DEVICE_IN_WIRED_HEADSET;
+ } else if (availableDeviceTypes & AUDIO_DEVICE_IN_USB_DEVICE) {
+ device = AUDIO_DEVICE_IN_USB_DEVICE;
+ } else if (availableDeviceTypes & AUDIO_DEVICE_IN_BUILTIN_MIC) {
+ device = AUDIO_DEVICE_IN_BUILTIN_MIC;
+ }
+ break;
+
+ case AUDIO_SOURCE_VOICE_COMMUNICATION:
+ // Allow only use of devices on primary input if in call and HAL does not support routing
+ // to voice call path.
+ if ((mPhoneState == AUDIO_MODE_IN_CALL) &&
+ (mAvailableOutputDevices.types() & AUDIO_DEVICE_OUT_TELEPHONY_TX) == 0) {
+ availableDeviceTypes = availablePrimaryInputDevices() & ~AUDIO_DEVICE_BIT_IN;
+ }
+
+ switch (mForceUse[AUDIO_POLICY_FORCE_FOR_COMMUNICATION]) {
+ case AUDIO_POLICY_FORCE_BT_SCO:
+ // if SCO device is requested but no SCO device is available, fall back to default case
+ if (availableDeviceTypes & AUDIO_DEVICE_IN_BLUETOOTH_SCO_HEADSET) {
+ device = AUDIO_DEVICE_IN_BLUETOOTH_SCO_HEADSET;
+ break;
+ }
+ // FALL THROUGH
+
+ default: // FORCE_NONE
+ if (availableDeviceTypes & AUDIO_DEVICE_IN_WIRED_HEADSET) {
+ device = AUDIO_DEVICE_IN_WIRED_HEADSET;
+ } else if (availableDeviceTypes & AUDIO_DEVICE_IN_USB_DEVICE) {
+ device = AUDIO_DEVICE_IN_USB_DEVICE;
+ } else if (availableDeviceTypes & AUDIO_DEVICE_IN_BUILTIN_MIC) {
+ device = AUDIO_DEVICE_IN_BUILTIN_MIC;
+ }
+ break;
+
+ case AUDIO_POLICY_FORCE_SPEAKER:
+ if (availableDeviceTypes & AUDIO_DEVICE_IN_BACK_MIC) {
+ device = AUDIO_DEVICE_IN_BACK_MIC;
+ } else if (availableDeviceTypes & AUDIO_DEVICE_IN_BUILTIN_MIC) {
+ device = AUDIO_DEVICE_IN_BUILTIN_MIC;
+ }
+ break;
+ }
+ break;
+
+ case AUDIO_SOURCE_VOICE_RECOGNITION:
+ case AUDIO_SOURCE_HOTWORD:
+ if (mForceUse[AUDIO_POLICY_FORCE_FOR_RECORD] == AUDIO_POLICY_FORCE_BT_SCO &&
+ availableDeviceTypes & AUDIO_DEVICE_IN_BLUETOOTH_SCO_HEADSET) {
+ device = AUDIO_DEVICE_IN_BLUETOOTH_SCO_HEADSET;
+ } else if (availableDeviceTypes & AUDIO_DEVICE_IN_WIRED_HEADSET) {
+ device = AUDIO_DEVICE_IN_WIRED_HEADSET;
+ } else if (availableDeviceTypes & AUDIO_DEVICE_IN_USB_DEVICE) {
+ device = AUDIO_DEVICE_IN_USB_DEVICE;
+ } else if (availableDeviceTypes & AUDIO_DEVICE_IN_BUILTIN_MIC) {
+ device = AUDIO_DEVICE_IN_BUILTIN_MIC;
+ }
+ break;
+ case AUDIO_SOURCE_CAMCORDER:
+ if (availableDeviceTypes & AUDIO_DEVICE_IN_BACK_MIC) {
+ device = AUDIO_DEVICE_IN_BACK_MIC;
+ } else if (availableDeviceTypes & AUDIO_DEVICE_IN_BUILTIN_MIC) {
+ device = AUDIO_DEVICE_IN_BUILTIN_MIC;
+ }
+ break;
+ case AUDIO_SOURCE_VOICE_DOWNLINK:
+ case AUDIO_SOURCE_VOICE_CALL:
+ if (availableDeviceTypes & AUDIO_DEVICE_IN_VOICE_CALL) {
+ device = AUDIO_DEVICE_IN_VOICE_CALL;
+ }
+ break;
+ case AUDIO_SOURCE_REMOTE_SUBMIX:
+ if (availableDeviceTypes & AUDIO_DEVICE_IN_REMOTE_SUBMIX) {
+ device = AUDIO_DEVICE_IN_REMOTE_SUBMIX;
+ }
+ break;
+ case AUDIO_SOURCE_FM_TUNER:
+ if (availableDeviceTypes & AUDIO_DEVICE_IN_FM_TUNER) {
+ device = AUDIO_DEVICE_IN_FM_TUNER;
+ }
+ break;
+ default:
+ ALOGW("getDeviceForInputSource() invalid input source %d", inputSource);
+ break;
+ }
+ ALOGV("getDeviceForInputSource()input source %d, device %08x", inputSource, device);
+ return device;
+}
+
+bool AudioPolicyManager::isVirtualInputDevice(audio_devices_t device)
+{
+ if ((device & AUDIO_DEVICE_BIT_IN) != 0) {
+ device &= ~AUDIO_DEVICE_BIT_IN;
+ if ((popcount(device) == 1) && ((device & ~APM_AUDIO_IN_DEVICE_VIRTUAL_ALL) == 0))
+ return true;
+ }
+ return false;
+}
+
+bool AudioPolicyManager::deviceDistinguishesOnAddress(audio_devices_t device) {
+ return ((device & APM_AUDIO_DEVICE_MATCH_ADDRESS_ALL & ~AUDIO_DEVICE_BIT_IN) != 0);
+}
+
+audio_io_handle_t AudioPolicyManager::getActiveInput(bool ignoreVirtualInputs)
+{
+ for (size_t i = 0; i < mInputs.size(); i++) {
+ const sp<AudioInputDescriptor> input_descriptor = mInputs.valueAt(i);
+ if ((input_descriptor->mRefCount > 0)
+ && (!ignoreVirtualInputs || !isVirtualInputDevice(input_descriptor->mDevice))) {
+ return mInputs.keyAt(i);
+ }
+ }
+ return 0;
+}
+
+uint32_t AudioPolicyManager::activeInputsCount() const
+{
+ uint32_t count = 0;
+ for (size_t i = 0; i < mInputs.size(); i++) {
+ const sp<AudioInputDescriptor> desc = mInputs.valueAt(i);
+ if (desc->mRefCount > 0) {
+ count++;
+ }
+ }
+ return count;
+}
+
+
+void AudioPolicyManager::initializeVolumeCurves()
+{
+ for (int i = 0; i < AUDIO_STREAM_CNT; i++) {
+ for (int j = 0; j < ApmGains::DEVICE_CATEGORY_CNT; j++) {
+ mStreams[i].mVolumeCurve[j] =
+ ApmGains::sVolumeProfiles[i][j];
+ }
+ }
+
+ // Check availability of DRC on speaker path: if available, override some of the speaker curves
+ if (mSpeakerDrcEnabled) {
+ mStreams[AUDIO_STREAM_SYSTEM].mVolumeCurve[ApmGains::DEVICE_CATEGORY_SPEAKER] =
+ ApmGains::sDefaultSystemVolumeCurveDrc;
+ mStreams[AUDIO_STREAM_RING].mVolumeCurve[ApmGains::DEVICE_CATEGORY_SPEAKER] =
+ ApmGains::sSpeakerSonificationVolumeCurveDrc;
+ mStreams[AUDIO_STREAM_ALARM].mVolumeCurve[ApmGains::DEVICE_CATEGORY_SPEAKER] =
+ ApmGains::sSpeakerSonificationVolumeCurveDrc;
+ mStreams[AUDIO_STREAM_NOTIFICATION].mVolumeCurve[ApmGains::DEVICE_CATEGORY_SPEAKER] =
+ ApmGains::sSpeakerSonificationVolumeCurveDrc;
+ mStreams[AUDIO_STREAM_MUSIC].mVolumeCurve[ApmGains::DEVICE_CATEGORY_SPEAKER] =
+ ApmGains::sSpeakerMediaVolumeCurveDrc;
+ mStreams[AUDIO_STREAM_ACCESSIBILITY].mVolumeCurve[ApmGains::DEVICE_CATEGORY_SPEAKER] =
+ ApmGains::sSpeakerMediaVolumeCurveDrc;
+ }
+}
+
+float AudioPolicyManager::computeVolume(audio_stream_type_t stream,
+ int index,
+ audio_io_handle_t output,
+ audio_devices_t device)
+{
+ float volume = 1.0;
+ sp<AudioOutputDescriptor> outputDesc = mOutputs.valueFor(output);
+ StreamDescriptor &streamDesc = mStreams[stream];
+
+ if (device == AUDIO_DEVICE_NONE) {
+ device = outputDesc->device();
+ }
+
+ volume = ApmGains::volIndexToAmpl(device, streamDesc, index);
+
+ // if a headset is connected, apply the following rules to ring tones and notifications
+ // to avoid sound level bursts in user's ears:
+ // - always attenuate ring tones and notifications volume by 6dB
+ // - if music is playing, always limit the volume to current music volume,
+ // with a minimum threshold at -36dB so that notification is always perceived.
+ const routing_strategy stream_strategy = getStrategy(stream);
+ if ((device & (AUDIO_DEVICE_OUT_BLUETOOTH_A2DP |
+ AUDIO_DEVICE_OUT_BLUETOOTH_A2DP_HEADPHONES |
+ AUDIO_DEVICE_OUT_WIRED_HEADSET |
+ AUDIO_DEVICE_OUT_WIRED_HEADPHONE)) &&
+ ((stream_strategy == STRATEGY_SONIFICATION)
+ || (stream_strategy == STRATEGY_SONIFICATION_RESPECTFUL)
+ || (stream == AUDIO_STREAM_SYSTEM)
+ || ((stream_strategy == STRATEGY_ENFORCED_AUDIBLE) &&
+ (mForceUse[AUDIO_POLICY_FORCE_FOR_SYSTEM] == AUDIO_POLICY_FORCE_NONE))) &&
+ streamDesc.mCanBeMuted) {
+ volume *= SONIFICATION_HEADSET_VOLUME_FACTOR;
+ // when the phone is ringing we must consider that music could have been paused just before
+ // by the music application and behave as if music was active if the last music track was
+ // just stopped
+ if (isStreamActive(AUDIO_STREAM_MUSIC, SONIFICATION_HEADSET_MUSIC_DELAY) ||
+ mLimitRingtoneVolume) {
+ audio_devices_t musicDevice = getDeviceForStrategy(STRATEGY_MEDIA, true /*fromCache*/);
+ float musicVol = computeVolume(AUDIO_STREAM_MUSIC,
+ mStreams[AUDIO_STREAM_MUSIC].getVolumeIndex(musicDevice),
+ output,
+ musicDevice);
+ float minVol = (musicVol > SONIFICATION_HEADSET_VOLUME_MIN) ?
+ musicVol : SONIFICATION_HEADSET_VOLUME_MIN;
+ if (volume > minVol) {
+ volume = minVol;
+ ALOGV("computeVolume limiting volume to %f musicVol %f", minVol, musicVol);
+ }
+ }
+ }
+
+ return volume;
+}
+
+status_t AudioPolicyManager::checkAndSetVolume(audio_stream_type_t stream,
+ int index,
+ audio_io_handle_t output,
+ audio_devices_t device,
+ int delayMs,
+ bool force)
+{
+
+ // do not change actual stream volume if the stream is muted
+ if (mOutputs.valueFor(output)->mMuteCount[stream] != 0) {
+ ALOGVV("checkAndSetVolume() stream %d muted count %d",
+ stream, mOutputs.valueFor(output)->mMuteCount[stream]);
+ return NO_ERROR;
+ }
+
+ // do not change in call volume if bluetooth is connected and vice versa
+ if ((stream == AUDIO_STREAM_VOICE_CALL &&
+ mForceUse[AUDIO_POLICY_FORCE_FOR_COMMUNICATION] == AUDIO_POLICY_FORCE_BT_SCO) ||
+ (stream == AUDIO_STREAM_BLUETOOTH_SCO &&
+ mForceUse[AUDIO_POLICY_FORCE_FOR_COMMUNICATION] != AUDIO_POLICY_FORCE_BT_SCO)) {
+ ALOGV("checkAndSetVolume() cannot set stream %d volume with force use = %d for comm",
+ stream, mForceUse[AUDIO_POLICY_FORCE_FOR_COMMUNICATION]);
+ return INVALID_OPERATION;
+ }
+
+ float volume = computeVolume(stream, index, output, device);
+ // unit gain if rerouting to external policy
+ if (device == AUDIO_DEVICE_OUT_REMOTE_SUBMIX) {
+ ssize_t index = mOutputs.indexOfKey(output);
+ if (index >= 0) {
+ sp<AudioOutputDescriptor> outputDesc = mOutputs.valueAt(index);
+ if (outputDesc->mPolicyMix != NULL) {
+ ALOGV("max gain when rerouting for output=%d", output);
+ volume = 1.0f;
+ }
+ }
+
+ }
+ // We actually change the volume if:
+ // - the float value returned by computeVolume() changed
+ // - the force flag is set
+ if (volume != mOutputs.valueFor(output)->mCurVolume[stream] ||
+ force) {
+ mOutputs.valueFor(output)->mCurVolume[stream] = volume;
+ ALOGVV("checkAndSetVolume() for output %d stream %d, volume %f, delay %d", output, stream, volume, delayMs);
+ // Force VOICE_CALL to track BLUETOOTH_SCO stream volume when bluetooth audio is
+ // enabled
+ if (stream == AUDIO_STREAM_BLUETOOTH_SCO) {
+ mpClientInterface->setStreamVolume(AUDIO_STREAM_VOICE_CALL, volume, output, delayMs);
+ }
+ mpClientInterface->setStreamVolume(stream, volume, output, delayMs);
+ }
+
+ if (stream == AUDIO_STREAM_VOICE_CALL ||
+ stream == AUDIO_STREAM_BLUETOOTH_SCO) {
+ float voiceVolume;
+ // Force voice volume to max for bluetooth SCO as volume is managed by the headset
+ if (stream == AUDIO_STREAM_VOICE_CALL) {
+ voiceVolume = (float)index/(float)mStreams[stream].mIndexMax;
+ } else {
+ voiceVolume = 1.0;
+ }
+
+ if (voiceVolume != mLastVoiceVolume && output == mPrimaryOutput) {
+ mpClientInterface->setVoiceVolume(voiceVolume, delayMs);
+ mLastVoiceVolume = voiceVolume;
+ }
+ }
+
+ return NO_ERROR;
+}
+
+void AudioPolicyManager::applyStreamVolumes(audio_io_handle_t output,
+ audio_devices_t device,
+ int delayMs,
+ bool force)
+{
+ ALOGVV("applyStreamVolumes() for output %d and device %x", output, device);
+
+ for (int stream = 0; stream < AUDIO_STREAM_CNT; stream++) {
+ if (stream == AUDIO_STREAM_PATCH) {
+ continue;
+ }
+ checkAndSetVolume((audio_stream_type_t)stream,
+ mStreams[stream].getVolumeIndex(device),
+ output,
+ device,
+ delayMs,
+ force);
+ }
+}
+
+void AudioPolicyManager::setStrategyMute(routing_strategy strategy,
+ bool on,
+ audio_io_handle_t output,
+ int delayMs,
+ audio_devices_t device)
+{
+ ALOGVV("setStrategyMute() strategy %d, mute %d, output %d", strategy, on, output);
+ for (int stream = 0; stream < AUDIO_STREAM_CNT; stream++) {
+ if (stream == AUDIO_STREAM_PATCH) {
+ continue;
+ }
+ if (getStrategy((audio_stream_type_t)stream) == strategy) {
+ setStreamMute((audio_stream_type_t)stream, on, output, delayMs, device);
+ }
+ }
+}
+
+void AudioPolicyManager::setStreamMute(audio_stream_type_t stream,
+ bool on,
+ audio_io_handle_t output,
+ int delayMs,
+ audio_devices_t device)
+{
+ StreamDescriptor &streamDesc = mStreams[stream];
+ sp<AudioOutputDescriptor> outputDesc = mOutputs.valueFor(output);
+ if (device == AUDIO_DEVICE_NONE) {
+ device = outputDesc->device();
+ }
+
+ ALOGVV("setStreamMute() stream %d, mute %d, output %d, mMuteCount %d device %04x",
+ stream, on, output, outputDesc->mMuteCount[stream], device);
+
+ if (on) {
+ if (outputDesc->mMuteCount[stream] == 0) {
+ if (streamDesc.mCanBeMuted &&
+ ((stream != AUDIO_STREAM_ENFORCED_AUDIBLE) ||
+ (mForceUse[AUDIO_POLICY_FORCE_FOR_SYSTEM] == AUDIO_POLICY_FORCE_NONE))) {
+ checkAndSetVolume(stream, 0, output, device, delayMs);
+ }
+ }
+ // increment mMuteCount after calling checkAndSetVolume() so that volume change is not ignored
+ outputDesc->mMuteCount[stream]++;
+ } else {
+ if (outputDesc->mMuteCount[stream] == 0) {
+ ALOGV("setStreamMute() unmuting non muted stream!");
+ return;
+ }
+ if (--outputDesc->mMuteCount[stream] == 0) {
+ checkAndSetVolume(stream,
+ streamDesc.getVolumeIndex(device),
+ output,
+ device,
+ delayMs);
+ }
+ }
+}
+
+void AudioPolicyManager::handleIncallSonification(audio_stream_type_t stream,
+ bool starting, bool stateChange)
+{
+ // if the stream pertains to sonification strategy and we are in call we must
+ // mute the stream if it is low visibility. If it is high visibility, we must play a tone
+ // in the device used for phone strategy and play the tone if the selected device does not
+ // interfere with the device used for phone strategy
+ // if stateChange is true, we are called from setPhoneState() and we must mute or unmute as
+ // many times as there are active tracks on the output
+ const routing_strategy stream_strategy = getStrategy(stream);
+ if ((stream_strategy == STRATEGY_SONIFICATION) ||
+ ((stream_strategy == STRATEGY_SONIFICATION_RESPECTFUL))) {
+ sp<AudioOutputDescriptor> outputDesc = mOutputs.valueFor(mPrimaryOutput);
+ ALOGV("handleIncallSonification() stream %d starting %d device %x stateChange %d",
+ stream, starting, outputDesc->mDevice, stateChange);
+ if (outputDesc->mRefCount[stream]) {
+ int muteCount = 1;
+ if (stateChange) {
+ muteCount = outputDesc->mRefCount[stream];
+ }
+ if (audio_is_low_visibility(stream)) {
+ ALOGV("handleIncallSonification() low visibility, muteCount %d", muteCount);
+ for (int i = 0; i < muteCount; i++) {
+ setStreamMute(stream, starting, mPrimaryOutput);
+ }
+ } else {
+ ALOGV("handleIncallSonification() high visibility");
+ if (outputDesc->device() &
+ getDeviceForStrategy(STRATEGY_PHONE, true /*fromCache*/)) {
+ ALOGV("handleIncallSonification() high visibility muted, muteCount %d", muteCount);
+ for (int i = 0; i < muteCount; i++) {
+ setStreamMute(stream, starting, mPrimaryOutput);
+ }
+ }
+ if (starting) {
+ mpClientInterface->startTone(AUDIO_POLICY_TONE_IN_CALL_NOTIFICATION,
+ AUDIO_STREAM_VOICE_CALL);
+ } else {
+ mpClientInterface->stopTone();
+ }
+ }
+ }
+ }
+}
+
+bool AudioPolicyManager::isInCall()
+{
+ return isStateInCall(mPhoneState);
+}
+
+bool AudioPolicyManager::isStateInCall(int state) {
+ return ((state == AUDIO_MODE_IN_CALL) ||
+ (state == AUDIO_MODE_IN_COMMUNICATION));
+}
+
+uint32_t AudioPolicyManager::getMaxEffectsCpuLoad()
+{
+ return MAX_EFFECTS_CPU_LOAD;
+}
+
+uint32_t AudioPolicyManager::getMaxEffectsMemory()
+{
+ return MAX_EFFECTS_MEMORY;
+}
+
+
+// --- EffectDescriptor class implementation
+
+status_t AudioPolicyManager::EffectDescriptor::dump(int fd)
+{
+ const size_t SIZE = 256;
+ char buffer[SIZE];
+ String8 result;
+
+ snprintf(buffer, SIZE, " I/O: %d\n", mIo);
+ result.append(buffer);
+ snprintf(buffer, SIZE, " Strategy: %d\n", mStrategy);
+ result.append(buffer);
+ snprintf(buffer, SIZE, " Session: %d\n", mSession);
+ result.append(buffer);
+ snprintf(buffer, SIZE, " Name: %s\n", mDesc.name);
+ result.append(buffer);
+ snprintf(buffer, SIZE, " %s\n", mEnabled ? "Enabled" : "Disabled");
+ result.append(buffer);
+ write(fd, result.string(), result.size());
+
+ return NO_ERROR;
+}
+
+
+// --- audio_policy.conf file parsing
+// TODO candidate to be moved to ConfigParsingUtils
+void AudioPolicyManager::loadHwModule(cnode *root)
+{
+ status_t status = NAME_NOT_FOUND;
+ cnode *node;
+ sp<HwModule> module = new HwModule(root->name);
+
+ node = config_find(root, DEVICES_TAG);
+ if (node != NULL) {
+ node = node->first_child;
+ while (node) {
+ ALOGV("loadHwModule() loading device %s", node->name);
+ status_t tmpStatus = module->loadDevice(node);
+ if (status == NAME_NOT_FOUND || status == NO_ERROR) {
+ status = tmpStatus;
+ }
+ node = node->next;
+ }
+ }
+ node = config_find(root, OUTPUTS_TAG);
+ if (node != NULL) {
+ node = node->first_child;
+ while (node) {
+ ALOGV("loadHwModule() loading output %s", node->name);
+ status_t tmpStatus = module->loadOutput(node);
+ if (status == NAME_NOT_FOUND || status == NO_ERROR) {
+ status = tmpStatus;
+ }
+ node = node->next;
+ }
+ }
+ node = config_find(root, INPUTS_TAG);
+ if (node != NULL) {
+ node = node->first_child;
+ while (node) {
+ ALOGV("loadHwModule() loading input %s", node->name);
+ status_t tmpStatus = module->loadInput(node);
+ if (status == NAME_NOT_FOUND || status == NO_ERROR) {
+ status = tmpStatus;
+ }
+ node = node->next;
+ }
+ }
+ loadGlobalConfig(root, module);
+
+ if (status == NO_ERROR) {
+ mHwModules.add(module);
+ }
+}
+
+// TODO candidate to be moved to ConfigParsingUtils
+void AudioPolicyManager::loadHwModules(cnode *root)
+{
+ cnode *node = config_find(root, AUDIO_HW_MODULE_TAG);
+ if (node == NULL) {
+ return;
+ }
+
+ node = node->first_child;
+ while (node) {
+ ALOGV("loadHwModules() loading module %s", node->name);
+ loadHwModule(node);
+ node = node->next;
+ }
+}
+
+// TODO candidate to be moved to ConfigParsingUtils
+void AudioPolicyManager::loadGlobalConfig(cnode *root, const sp<HwModule>& module)
+{
+ cnode *node = config_find(root, GLOBAL_CONFIG_TAG);
+
+ if (node == NULL) {
+ return;
+ }
+ DeviceVector declaredDevices;
+ if (module != NULL) {
+ declaredDevices = module->mDeclaredDevices;
+ }
+
+ node = node->first_child;
+ while (node) {
+ if (strcmp(ATTACHED_OUTPUT_DEVICES_TAG, node->name) == 0) {
+ mAvailableOutputDevices.loadDevicesFromName((char *)node->value,
+ declaredDevices);
+ ALOGV("loadGlobalConfig() Attached Output Devices %08x",
+ mAvailableOutputDevices.types());
+ } else if (strcmp(DEFAULT_OUTPUT_DEVICE_TAG, node->name) == 0) {
+ audio_devices_t device = (audio_devices_t)ConfigParsingUtils::stringToEnum(
+ sDeviceNameToEnumTable,
+ ARRAY_SIZE(sDeviceNameToEnumTable),
+ (char *)node->value);
+ if (device != AUDIO_DEVICE_NONE) {
+ mDefaultOutputDevice = new DeviceDescriptor(String8("default-output"), device);
+ } else {
+ ALOGW("loadGlobalConfig() default device not specified");
+ }
+ ALOGV("loadGlobalConfig() mDefaultOutputDevice %08x", mDefaultOutputDevice->mDeviceType);
+ } else if (strcmp(ATTACHED_INPUT_DEVICES_TAG, node->name) == 0) {
+ mAvailableInputDevices.loadDevicesFromName((char *)node->value,
+ declaredDevices);
+ ALOGV("loadGlobalConfig() Available InputDevices %08x", mAvailableInputDevices.types());
+ } else if (strcmp(SPEAKER_DRC_ENABLED_TAG, node->name) == 0) {
+ mSpeakerDrcEnabled = ConfigParsingUtils::stringToBool((char *)node->value);
+ ALOGV("loadGlobalConfig() mSpeakerDrcEnabled = %d", mSpeakerDrcEnabled);
+ } else if (strcmp(AUDIO_HAL_VERSION_TAG, node->name) == 0) {
+ uint32_t major, minor;
+ sscanf((char *)node->value, "%u.%u", &major, &minor);
+ module->mHalVersion = HARDWARE_DEVICE_API_VERSION(major, minor);
+ ALOGV("loadGlobalConfig() mHalVersion = %04x major %u minor %u",
+ module->mHalVersion, major, minor);
+ }
+ node = node->next;
+ }
+}
+
+// TODO candidate to be moved to ConfigParsingUtils
+status_t AudioPolicyManager::loadAudioPolicyConfig(const char *path)
+{
+ cnode *root;
+ char *data;
+
+ data = (char *)load_file(path, NULL);
+ if (data == NULL) {
+ return -ENODEV;
+ }
+ root = config_node("", "");
+ config_load(root, data);
+
+ loadHwModules(root);
+ // legacy audio_policy.conf files have one global_configuration section
+ loadGlobalConfig(root, getModuleFromName(AUDIO_HARDWARE_MODULE_ID_PRIMARY));
+ config_free(root);
+ free(root);
+ free(data);
+
+ ALOGI("loadAudioPolicyConfig() loaded %s\n", path);
+
+ return NO_ERROR;
+}
+
+void AudioPolicyManager::defaultAudioPolicyConfig(void)
+{
+ sp<HwModule> module;
+ sp<IOProfile> profile;
+ sp<DeviceDescriptor> defaultInputDevice =
+ new DeviceDescriptor(String8("builtin-mic"), AUDIO_DEVICE_IN_BUILTIN_MIC);
+ mAvailableOutputDevices.add(mDefaultOutputDevice);
+ mAvailableInputDevices.add(defaultInputDevice);
+
+ module = new HwModule("primary");
+
+ profile = new IOProfile(String8("primary"), AUDIO_PORT_ROLE_SOURCE, module);
+ profile->mSamplingRates.add(44100);
+ profile->mFormats.add(AUDIO_FORMAT_PCM_16_BIT);
+ profile->mChannelMasks.add(AUDIO_CHANNEL_OUT_STEREO);
+ profile->mSupportedDevices.add(mDefaultOutputDevice);
+ profile->mFlags = AUDIO_OUTPUT_FLAG_PRIMARY;
+ module->mOutputProfiles.add(profile);
+
+ profile = new IOProfile(String8("primary"), AUDIO_PORT_ROLE_SINK, module);
+ profile->mSamplingRates.add(8000);
+ profile->mFormats.add(AUDIO_FORMAT_PCM_16_BIT);
+ profile->mChannelMasks.add(AUDIO_CHANNEL_IN_MONO);
+ profile->mSupportedDevices.add(defaultInputDevice);
+ module->mInputProfiles.add(profile);
+
+ mHwModules.add(module);
+}
+
+audio_stream_type_t AudioPolicyManager::streamTypefromAttributesInt(const audio_attributes_t *attr)
+{
+ // flags to stream type mapping
+ if ((attr->flags & AUDIO_FLAG_AUDIBILITY_ENFORCED) == AUDIO_FLAG_AUDIBILITY_ENFORCED) {
+ return AUDIO_STREAM_ENFORCED_AUDIBLE;
+ }
+ if ((attr->flags & AUDIO_FLAG_SCO) == AUDIO_FLAG_SCO) {
+ return AUDIO_STREAM_BLUETOOTH_SCO;
+ }
+ if ((attr->flags & AUDIO_FLAG_BEACON) == AUDIO_FLAG_BEACON) {
+ return AUDIO_STREAM_TTS;
+ }
+
+ // usage to stream type mapping
+ switch (attr->usage) {
+ case AUDIO_USAGE_MEDIA:
+ case AUDIO_USAGE_GAME:
+ case AUDIO_USAGE_ASSISTANCE_NAVIGATION_GUIDANCE:
+ return AUDIO_STREAM_MUSIC;
+ case AUDIO_USAGE_ASSISTANCE_ACCESSIBILITY:
+ if (isStreamActive(AUDIO_STREAM_ALARM)) {
+ return AUDIO_STREAM_ALARM;
+ }
+ if (isStreamActive(AUDIO_STREAM_RING)) {
+ return AUDIO_STREAM_RING;
+ }
+ if (isInCall()) {
+ return AUDIO_STREAM_VOICE_CALL;
+ }
+ return AUDIO_STREAM_ACCESSIBILITY;
+ case AUDIO_USAGE_ASSISTANCE_SONIFICATION:
+ return AUDIO_STREAM_SYSTEM;
+ case AUDIO_USAGE_VOICE_COMMUNICATION:
+ return AUDIO_STREAM_VOICE_CALL;
+
+ case AUDIO_USAGE_VOICE_COMMUNICATION_SIGNALLING:
+ return AUDIO_STREAM_DTMF;
+
+ case AUDIO_USAGE_ALARM:
+ return AUDIO_STREAM_ALARM;
+ case AUDIO_USAGE_NOTIFICATION_TELEPHONY_RINGTONE:
+ return AUDIO_STREAM_RING;
+
+ case AUDIO_USAGE_NOTIFICATION:
+ case AUDIO_USAGE_NOTIFICATION_COMMUNICATION_REQUEST:
+ case AUDIO_USAGE_NOTIFICATION_COMMUNICATION_INSTANT:
+ case AUDIO_USAGE_NOTIFICATION_COMMUNICATION_DELAYED:
+ case AUDIO_USAGE_NOTIFICATION_EVENT:
+ return AUDIO_STREAM_NOTIFICATION;
+
+ case AUDIO_USAGE_UNKNOWN:
+ default:
+ return AUDIO_STREAM_MUSIC;
+ }
+}
+
+bool AudioPolicyManager::isValidAttributes(const audio_attributes_t *paa) {
+ // has flags that map to a strategy?
+ if ((paa->flags & (AUDIO_FLAG_AUDIBILITY_ENFORCED | AUDIO_FLAG_SCO | AUDIO_FLAG_BEACON)) != 0) {
+ return true;
+ }
+
+ // has known usage?
+ switch (paa->usage) {
+ case AUDIO_USAGE_UNKNOWN:
+ case AUDIO_USAGE_MEDIA:
+ case AUDIO_USAGE_VOICE_COMMUNICATION:
+ case AUDIO_USAGE_VOICE_COMMUNICATION_SIGNALLING:
+ case AUDIO_USAGE_ALARM:
+ case AUDIO_USAGE_NOTIFICATION:
+ case AUDIO_USAGE_NOTIFICATION_TELEPHONY_RINGTONE:
+ case AUDIO_USAGE_NOTIFICATION_COMMUNICATION_REQUEST:
+ case AUDIO_USAGE_NOTIFICATION_COMMUNICATION_INSTANT:
+ case AUDIO_USAGE_NOTIFICATION_COMMUNICATION_DELAYED:
+ case AUDIO_USAGE_NOTIFICATION_EVENT:
+ case AUDIO_USAGE_ASSISTANCE_ACCESSIBILITY:
+ case AUDIO_USAGE_ASSISTANCE_NAVIGATION_GUIDANCE:
+ case AUDIO_USAGE_ASSISTANCE_SONIFICATION:
+ case AUDIO_USAGE_GAME:
+ case AUDIO_USAGE_VIRTUAL_SOURCE:
+ break;
+ default:
+ return false;
+ }
+ return true;
+}
+
+}; // namespace android